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.     

Robustness of spatial average equalization: a statistical reverberation model approach

Traditionally, multiple listener room equalization is performed to improve sound quality at all listeners, during audio playback, in a multiple listener environment (e.g., movie theaters, automobiles, etc.). A typical way of doing multiple listener equalization is through spatial averaging, where the room responses are averaged spatially between positions and an inverse equalization filter is found from the spatially averaged result. However, the equalization performance, will be affected if there is a mismatch between the position of the microphones (which are used for measuring the room responses for designing the equalization filter) and the actual center of listener head position (during playback). In this paper, we will present results on the effects of microphone-listener mismatch on spatial average equalization performance. The results indicate that, for the analyzed rectangular configuration, the region of effective equalization depends on (i) the distance of a listener from the source, (ii) the amount of mismatch between the responses, and (iii) the frequency of the audio signal. We also present some convergence analysis to interpret the results.     

Robustness of multichannel equalization in an acoustic reverberant environment

A theoretical framework is established, for the robustness of multichannel sound equalization in reverberant environments. Using results from statistical room acoustics, a closed-form expression is derived that predicts the degradation in performance of an equalization system as the sound source moves from its nominal position inside the enclosure. The presented analysis also provides means of identifying the performance bounds that can be expected when using such a system in an actual room. Using extensive computer simulations, the effect of physical parameters such as the relative positions of the source and the receivers, as well as effects of different design parameters are investigated. Based on the conditions imposed by these parameters, it is shown that, depending on the array geometry and the exact form of the equalizers, slight performance gains can be expected as the number of receivers is increased.     

Efficient equalization of multi-exciter distributed mode loudspeakers

An efficient digital equalization method is applied successfully to the problem of spectral equalization of multi-exciter distributed mode loudspeakers (DML). It is based on a chain of second-order sections of infinite impulse response parametric filters with very low computational cost. The method compensates for the measured multi-exciter DML response in order to achieve a desired frequency response. The sound radiation of these flat loudspeakers is a complex superposition of excited modes that vary strongly with frequency. Therefore, the characteristic multi-exciter DML spectrum is very irregular and is equalized with the method presented here for a natural, uncolored response. In multichannel systems, such as wave field synthesis (WFS), the use of efficient filters to equalize a large amount of drivers is an advantageous approach. The equalization process has been applied to two multi-exciter DML prototypes, comprising three and five exciters per panel. Both panel and exciter equalization have been addressed, which consequences on the filtered responses are discussed. Finally, some subjective assessments are carried out to optimize the order of the filter while maintaining the perceived quality of the equalization.     

Spectral-energy characteristics of sound fields scattered by fine-structured inhomogeneities of sound velocity into the geometrical shadow zone

In analyzing experimental results obtained with explosive sources in the tropical zone of the Indian Ocean, a good agreement was obtained for spectral-energy characteristics of signals observed in the first geometrical shadow zone with computer calculations of the sound field scattered by fine structure inhomogeneities of the fractal type. From the comparison of the results of calculating the frequency characteristics of sound fields in the shadow zone by the wave code and by the method combining ray acoustics with the wave theory of sound scattering, it was found that both methods are appropriate for describing the real processes of scattering and propagation of sound in the ocean with fine-structured stratification and that these methods can be used for solving inverse problems.     

Equalization filter design for downfiring flat television speakers

Many loudspeakers in newer flat television systems are mounted on the bottom of the television, with their diaphragms facing downward, so as to be hidden inside the TV frame. This kind of loudspeaker installation, called downfiring, induces relatively large reflections from the walls of a room. The increased reflections inevitably change the impulse response of the loudspeakers, which leads to a perceptible distortion in sound quality. In this study, an equalization procedure to resolve the distortion due to downfiring loudspeakers is presented. The change in the early-arriving sound from the room reflections was analyzed, and the specific experimental environment was set up to measure the early reflections for designing a single-channel equalization filter. The inverse technique for the single-input multiple-output (SIMO) system was applied such that both the magnitude and phase responses from downfiring loudspeakers can be equalized at multiple listener positions. We also investigated the possible equalization range in space and time, for which the single-channel filter can work effectively. The performance of the filter designed for the equalization range was then demonstrated by experiments. The experiments were performed using a flat television mock-up installed in a reflective environment. For the performance evaluation of the proposed method, we employed two different measures that can represent the magnitude and phase distortions: the mean squared error of the magnitude and an energy decay curve. The experimental results demonstrate that the single-channel filter can reduce the early reflections from a downfiring loudspeaker by a considerable amount.     

复杂声学环境中人耳附近空间有源降噪研究综述

复杂声学环境中人耳附近空间降噪是有源噪声控制研究的重要课题,目前采用的主要方法为有源降噪头靠(AHR)和虚拟声屏障(VSB).本文简述AHR与VSB的发展历史和研究现状,介绍其物理原理和设计方法,评述其在实际应用中的优缺点,讨论了目前存在的问题与未来相关的研究方向.已有理论、数值仿真和实验研究验证了相关技术在人耳附近空间产生静区的可行性. AHR系统需要较少控制源,系统相对简单易实现,但静区范围较小,结合虚拟传声器技术和人头跟踪技术后可实现随人头移动的静区,降噪频率可达中高频; VSB产生的静区范围较大,但控制源个数较多,系统复杂和成本高,可通过代价函数和控制源优化,以及主被动混合控制技术来提高有效降噪频率范围和减少控制源个数.     

On the externalization of virtual sound images in headphone reproduction: a Wiener filter approach

In-head localization of sound images is a critical problem in headphone reproduction. The paper investigates the degree of externalization in terms of the distance of auditory images for various synthesis and reproduction cases. An effective binaural headphone system was constructed by way of binaural synthesis using head-related impulse responses and individual headphone equalization using Wiener filter theory. The headphone system designed had an average reproduction performance error of 2.4% for five subjects with a random noise input, and was used to perform some subjective tests with a set of virtual sources equally spaced and distanced from the center of each subject's head in the horizontal plane. The effects of individual and nonindividual binaural syntheses and those of equalized and nonequalized reproductions were separately investigated. In the tests, each subject was instructed to indicate the distance of auditory images. The results obtained demonstrate that individual equalization is important for externalization, and individual synthesis is important for consistent distance perception. Thus, a combined use of both individual equalization and individual synthesis resulted in externalized sound images of a consistent distance.     

Researches on the distribution law of vector sound field in elastic wedge bottom

The method based on elastic parabolic equation method for calculating the sound vector field has been studied.The vector field in water and corresponding seismic wave field had been calculated for infra-sound in oceanic environment with elastic wedge bottom.The effects on sound field distribution for different frequency and depth of sound source had been researched,result shows that there is sound energy leakage into the bottom,the position where leakage occurred can be determined by the ratio of the ocean depth to the wavelength,as compared with normal mode theory.     

Controlled sound field with a dual layer loudspeaker array

Controlled sound interference has been extensively investigated using a prototype dual layer loudspeaker array comprised of 16 loudspeakers. Results are presented for measures of array performance such as input signal power, directivity of sound radiation and accuracy of sound reproduction resulting from the application of conventional control methods such as minimization of error in mean squared pressure, maximization of energy difference and minimization of weighted pressure error and energy. Procedures for selecting the tuning parameters have also been introduced. With these conventional concepts aimed at the production of acoustically bright and dark zones, all the control methods used require a trade-off between radiation directivity and reproduction accuracy in the bright zone. An alternative solution is proposed which can achieve better performance based on the measures presented simultaneously by inserting a low priority zone named as the “gray” zone. This involves the weighted minimization of mean-squared errors in both bright and dark zones together with the gray zone in which the minimization error is given less importance. This results in the production of directional bright zone in which the accuracy of sound reproduction is maintained with less required input power. The results of simulations and experiments are shown to be in excellent agreement.     

深海海底反射区声场的相干结构及其有源声呐估距

有源声呐在探测深海海底反射区的目标时,由于声线大掠射角弯曲且声速沿声线传播路径不断变化,造成了常规估距方法产生较大的误差。有效声速法是减小常规估距方法误差的有效途径,但由于需要预先计算空间每一位置点的"声线时延有效声速"对,复杂度高,实时性差。针对有效声速法的实时性问题,本文基于深海海底反射区声场的相干结构,提出一种改进的有效声速估距方法.首先指出深海声场能量沿声源出射角强弱相间变化及其引起的海底反射区离散声呐可探测区现象,并利用深海近水面声源的声线干涉效应解释了该现象的物理机理,建立了声呐可探测区与高能量声线的量化关系。在此基础上,计算声呐可探测区边界位置的"声线时延-有效声速"对,并线性拟合出可探测区所有位置点对应的值。经仿真验证,该方法与传统的有效声速法均可实现对常规估距方法估距误差的有效校正。虽然该方法估距精度较传统的有效声速法略有增大,但计算复杂度和计算时间显著减小,实时性好,具有良好的工程应用前景。      

Improved Approach for the Maximum Entropy Deconvolution Problem

The probability density function (pdf) valid for the Gaussian case is often applied for describing the convolutional noise pdf in the blind adaptive deconvolution problem, although it is known that it can be applied only at the latter stages of the deconvolution process, where the convolutional noise pdf tends to be approximately Gaussian. Recently, the deconvolutional noise pdf was approximated with the Edgeworth Expansion and with the Maximum Entropy density function for the 16 Quadrature Amplitude Modulation (QAM) input but no equalization performance improvement was seen for the hard channel case with the equalization algorithm based on the Maximum Entropy density function approach for the convolutional noise pdf compared with the original Maximum Entropy algorithm, while for the Edgeworth Expansion approximation technique, additional predefined parameters were needed in the algorithm. In this paper, the Generalized Gaussian density (GGD) function and the Edgeworth Expansion are applied for approximating the convolutional noise pdf for the 16 QAM input case, with no need for additional predefined parameters in the obtained equalization method. Simulation results indicate that improved equalization performance is obtained from the convergence time point of view of approximately 15,000 symbols for the hard channel case with our new proposed equalization method based on the new model for the convolutional noise pdf compared to the original Maximum Entropy algorithm. By convergence time, we mean the number of symbols required to reach a residual inter-symbol-interference (ISI) for which reliable decisions can be made on the equalized output sequence.     

An active noise barrier with unidirectional secondary sources

An active noise barrier with unidirectional secondary sources is investigated in this paper, where the unidirectional secondary source consists of two closely located loudspeakers with pre-adjusted phase difference. The secondary sound field of the unidirectional sources is adjusted to maximally match the primary sound field in the shadow zone behind the barrier. It is shown both numerically and experimentally that the noise reduction performance of the active noise barrier can be improved remarkably by replacing monopoles with the unidirectional sources. The mechanism for the improvement is also investigated.     

一种强指向性基阵的设计及声场计算

提出了一种新的基阵:将90个小圆环分布于半径不同的5个同心圆周上,对每个小圆环采用了抛物型的振速加权。通过改变阵元尺寸、间距等参数研究了声场特征的变化。计算结果表明,该基阵模型具有非常高的指向性和抑制旁瓣的能力。通过优化阵元组合方式和参数,可获得最佳的声压分布效果。     

The study of underwater acoustic communication technology based-on the acoustic vector sensor

Underwater acoustic (UWA) communication based on an acoustic vector sensor is studied. The method of joint weighted sound pressure and velocity processing is used in phase modulation high-speed UWA communication system combined with coherent demodulation and adaptive equalization algorithm to demodulate and decode. Whereas the sound intensity could be used instead of pressure for frequency decoding in frequency modulation UWA communication system. The results of theory analysis, simulation calculations and lake trials have shown that either in phase modulation or in frequency modulation UWA communication system, the processing gain can be evidently increased, so that the BER (bit error rate) can be effectively reduced and the telemetry distance can be enlarged by using the acoustic vector sensor.     

深海海底山环境下声传播水平折射效应研究

声波在深海海底山环境中传播时,海底山会对声传播产生重要影响.2016年在南海深海进行了一次海底山环境下的声传播实验,观测到了由海底山引起的三维声传播效应,本文利用BELLHOP射线理论解释了海底山环境下的三维声传播机理.结果表明：声波在传播过程中与海底山作用后破坏了深海会聚区结构,导致传播损失增大,在海底山后形成具有明显边界的声水平折射区,利用二维声传播模型无法解释实验现象,海底山后声水平折射区实验测量的声场结构与N×2D模型计算结果存在明显差异,实验的传播损失比N×2D模型计算结果大10 dB.通过三维射线模型分析N×2D模型计算结果与实验结果存在明显差异产生的原因,发现由于声波水平折射作用,部分声线无法到达接收器,使得三维声传播效应对海底山后一定角度范围内声场影响较为明显.因此,深海海底山会引起明显的三维水平折射效应,应在水下目标探测和定位等应用中给予重视.     

Propagation of sound from a monopole source above an impedance-backed porous layer

In this article, the propagation of sound from a monopole source above an impedance-backed porous layer is examined. The sound fields can be expressed in an integral form that is amenable to further analysis. A standard method of steepest descents is applied to evaluate the integral where the method of pole subtraction is needed to obtain a uniform asymptotic solution for the sound field above the plane surface. To obtain a numerical solution, the location of the pole was determined numerically by means of the Newton-Raphson method. Based on the pole location, the sound fields can then be calculated numerically. It has been demonstrated that the use of a plane wave reflection coefficient to calculate the sound fields is a special case of the asymptotic formula when the pole is located further away from the saddle point.     

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。     

A time-selective technique for free-field reciprocity calibration of condenser microphones

In normal practice, microphones are calibrated in a closed coupler where the sound pressure is uniformly distributed over the diaphragm. Alternatively, microphones can be placed in a free field, although in that case the distribution of sound pressure over the diaphragm will change as a result of the diffraction of the body of the microphone, and thus, its sensitivity will change. In the two cases, a technique based on the reciprocity theorem can be applied for obtaining the absolute sensitivity either under uniform pressure or free-field conditions. In this paper, signal-processing techniques are considered that improve the accuracy of the free-field calibration method. In particular, a fast Fourier transform (FFT)-based time-selective technique for removing undesired reflections from the walls of the measurement chamber has been developed and applied to the electric transfer impedance function between two microphones. The acoustic centers of the microphones have been determined from the "cleaned" transfer impedance values. Then, the complex free-field sensitivities of the microphones have been calculated. The resulting complex sensitivities and acoustic centers have proved to be in good agreement with previously published data, and this confirms the reliability of the time-selective technique, even in nonanechoic environments. Furthermore, the obtained results give a new reference for further comparisons, because they cover a frequency range with an accuracy that has not been obtained by previously published data.     

深海海底反射会聚区声传播特性

在深海声道条件下,海水折射效应会使得声场出现会聚效应;在不完全声道条件下,深海海底对声场具有重要影响.利用在中国南海海域收集到的一次深海声传播实验数据,研究了深海不完全声道环境下的海底反射对声传播的影响.实验观测到不同于深海会聚区的海底反射会聚现象,在直达声区范围内的海底地形隆起可导致海底反射会聚区提前形成,并使得部分影区的声强明显提高.由于不平坦海底和海面的反射破坏了完全声道环境下的会聚区结构,在60 km范围内存在两个海底反射会聚区,会聚区增益可达10 dB以上,同时在11 km附近的影区和51 km附近形成高声强区域.当接收深度与声源深度相同时,第二会聚区的增益高于第一会聚区.在第一会聚区内,随着接收深度的增加,声线到达结构趋于复杂,多途效应更加明显.使用抛物方程数值分析结合射线理论对深海海底反射会聚区现象产生的物理原因进行了分析解释.研究结果对于声纳在深海复杂环境下的性能分析具有重要的指导意义.