次级声源优化布放的局部空间有源降噪

针对区域有源降噪问题,为获得更优降噪效果,根据实际次级通路传递函数,提出次级声源优化布放的有源控制系统并详细比较了两种次级声源优化布放算法与次级声源均匀布放的实际降噪效果。应用的第一种次级声源优化算法是l2范数约束的约束匹配追踪算法,第二种次级声源优化算法是l1范数约束的稀疏正则化方法。在全消声室中利用扬声器线阵进行多通道有源降噪实验研究,实验结果表明,在200~1000 Hz,次级声源优化布放的控制系统的平均降噪量比次级声源均匀布放的控制系统的平均降噪量多5 dB左右;在1100~1900 Hz,次级声源优化布放的控制系统的平均降噪量比次级声源均匀布放的控制系统的平均降噪量多11~13 dB左右,次级声源优化布放的控制系统的降噪量分布更加均匀且次级声源输出能量更小。此外,两种优化算法中,稀疏正则化方法的降噪效果更佳。     

平板扬声器用于结构声有源噪声控制实验研究

对NXT技术的分布式平板扬声器和集中参数扬声器的频响和指向性等进行了对比测量，通过对组合平板扬声器的声场研究及有源消声实验，证明了平板扬声器用于有源噪声控制的可行性。综合平板扬声器和集中参数扬声器的物理特点和声场特性，根据基于平面声源有源噪声控制的已有的理论研究结果和本文的实验结果，可以认为，NXT平板扬声器适合作有源噪声控制的次级源，有利于有源噪声控制的工程推广应用。     

管道噪声的自适应抵消及其计算机模拟

本文提出一种采用自适应抵消的管道噪声有源降噪系统，自适应滤波器为有限冲激响应的横向结构，系统中采用了辅助滤波器和补偿滤波器，它们保证了最小均方算法的正确收敛，消除了声反馈效应，计算机模拟证实了该系统的可行性，通过对具有不同物理参数的管道的计算机模拟，得到了这些参数的变化对总降噪量的影响，这些参数包括温度、气流速度、管内的吸声情形、次级声源的Q值及次级声源到检测传声器的距离，模拟的结果有助于实际管道有源降噪的系统的设计。     

几种管道有源降噪器控制系统频率响应函数的确定

管道有源降噪器的控制系统根据传声器检测到的噪声信号,经过加工处理,产生激励次级声源的信号,使其产生的声波在下游与原来的噪声相抵消。控制系统应具备的频率响应函数取决于次级声源及检测传声器的配置,及其本身的频率响应。本文推导了在常用的几种不同次级声源和检测传声器配置下,所需的控制系统的频率响应函数;并介绍了一种测量传声器及扬声器在管道中的频率响应的方法。它们为控制系统的设计提供了依据。     

用于开口声辐射控制的虚拟声屏障实现 方法及机理研究*

建筑物通常留有开口以便人员物料的进出及室内的自然通风采光,但这些开口也是噪声传播的途径。传统被动噪声控制方法需要将开口封闭,且对低频噪声的控制效果不好,故引入有源噪声控制技术降低室内声源通过开口的声辐射。基于惠更斯原理,均布开口的次级源和误差传声器构成的平面型虚拟声屏障可以实现对开口声辐射的有效控制,数值仿真和实验已证明其有效性。将次级源安装在开口边界更有利于保留开口的功能且方便实际安装,但这样的单层边界虚拟声屏障降噪效果存在上限,仅能在低频段实现全局控制。和单层边界次级源相比,双层边界次级源可显著提高降噪量和有效降噪频率上限。该文回顾了开口声辐射有源控制的相关工作,并讨论了未来可能的研究方向。     

关于单极子次级声源管道有源降噪能量机制的理论分析

本文通过对单极子次级声源管道有源降噪中声场能量的再分布与初、次级声源的工作情况的定量分析与讨论,说明了有源降噪的三种能量机制中的多极子能量存贮机制,总是同能量转移或吸收机制中的一种同时存在的,哪一种机制起主导作用取决于初、次级声源之间的距离.     

面向目标的定向声辐射技术及其应用研究

面向目标的定向声辐射是指利用声源特性和阵列信号处理技术,将声波传送至目标方向或区域的声场控制方法。本文介绍了定向声辐射技术的原理和进展,重点阐述了普通扬声器阵列和参量阵扬声器这两种声源形式的理论、算法和应用研究。其中,针对扬声器阵列,主要分析了基于声能量对比度控制的声聚焦技术;针对参量阵扬声器,综述了利用非线性效应产生高指向性可听声的计算方法、信号预处理和应用等进展。面向目标的定向声辐射技术的研究对实现智能声场控制有着重要意义。     

壁面分布次级声源对管道中高阶模式声波的有源控制

针对管道中高阶模式声波的有源控制问题,探究了管道壁面布放次级声源的控制机制。首先根据管道声场模式理论和有源控制准则,推导了无限长管道内次级声源激励声场实现控制的必要条件,然后论证了次级声源沿管道壁面分布的可控性,并提出将空间分布离散化后的控制策略;最后通过数值仿真验证了沿壁面分布的有效性。结果表明,当次级声源在管道壁面的周向分布满足空间采样定律且沿轴向分布具有足够长度时,可以取得显著的有源控制效果.      

次级源近场和管壁弹性对充液管路有源消声性能的影响EI北大核心CSCD

建立了含次级源结构的充液直管有源消声系统数值模型,重点分析了声激励下次级源近场和管壁弹性对有源消声性能的影响。结果表明:次级源近场为非均匀声场,误差点位于该区域时部分频点控制效果较差甚至放大,而处于声场均匀区域时可使降噪量提高10 dB以上,增加误差点数量可使绝大多数频点的降噪量提高5 dB以上;管壁弹性使次级源与管壁间的耦合较强,非对称分布的次级源容易激起管壁振动,导致降噪谷值的出现,采用对称分布的次级源可显著提升控制效果;增加次级源数量能够提高系统的有源无源复合控制效果,但使得管内声场变得复杂,多次级源模型的有源消声效果随频率升高而有所降低。     

有源声屏障中误差传感器的位置优化

有源声屏障利用有源控制系统提高声屏障低频段的降噪效果。有源控制系统中误差传感器的位置对整个系统的降噪效果有较大的影响。通过数值模拟和实验研究误差传感器的位置优化问题,得出了有源控制系统中误差传感器摆放位置的两条结论:(1)所介绍的三种摆放中,误差传感器的位置在次级声源的正上方时,有源控制系统在屏障后方声影区引入的新增插入损失最好,特别是对于距离屏障较远的区域;(2)当误差传感器的位置在次级声源的正上方时,误差传感器与次级声源间的距离存在一个最优距离使得屏障后方声影区的衍射声得到最好的降低。     

GA-based optimization of a MIMO ANC system considering coupling of secondary sources in a telephone kiosk

In this paper a multi-input, multi-output (MIMO) active noise control system with the aim of global reduction of broadband noise in a telephone kiosk is addressed. The model selected for this optimization problem is the acoustic environment of an enclosure taking into account the effect of coupling of secondary sources used for control purpose. This optimization involves finding the best locations for loudspeakers and microphones inside the enclosure as well as optimizing the control signals considering secondary source coupling.Previous results show that in order to be able to reduce acoustic noise globally inside the enclosure, the frequency range of 50-300 Hz must be selected for control purpose. The mean of acoustic potential energy of the enclosure, when excited in this frequency range, is adopted as a performance measure. This performance index is penalized with the power of the signal required to excite secondary loudspeakers, in order to avoid placements that may need high voltage power amplifier for a desired performance. To find the solution of this problem, i.e. the global minimum of the performance index, several genetic algorithms are proposed and compared. In order to attain the best achievable performance in reaching the global minimum, the parameters of these genetic algorithms are tuned, and used for optimization purpose. Numerical simulations of the acoustical potential energy as well as the sound pressure at different heights of the kiosk, when active noise control (ANC) system operates, confirm the optimality of the locations proposed by the genetic algorithm.     

Binaural active noise control using parametric array loudspeakers

This paper reports the binaural active noise control (ANC) system developed to deal with factory noise. The control points are located in the vicinity of the left and right ears of a worker sitting along the production line. Due to the complicated safety requirements in the factory, secondary sources and error microphones are not allowed to be placed near the worker. Therefore, the proposed ANC system employs the feedforward structure and adopts the parametric array loudspeakers (PALs) as the secondary sources. The PAL is a type of directional loudspeaker that generates a much narrower sound field as compared to the conventional loudspeaker. Once the proposed ANC system has been trained offline, the error microphones can be removed. The performance of the binaural ANC system is successfully demonstrated based on a digital signal processor (DSP) implementation.     

Active control of the volume acquisition noise in functional magnetic resonance imaging: method and psychoacoustical evaluation.

Functional magnetic resonance imaging (fMRI) provides a noninvasive tool for observing correlates of neural activity in the brain while a subject listens to sound. However, intense acoustic noise is generated in the process of capturing MR images. This noise stimulates the auditory nervous system, limiting the dynamic range available for displaying stimulus-driven activity. The noise is potentially damaging to hearing and is distracting for the subject. In an active noise control (ANC) system, a reference sample of a noise is processed to form a sound which adds destructively with the noise at the listener's ear. We describe an implementation of ANC in the electromagnetically hostile and physically compact MRI scanning environment. First, a prototype system was evaluated psychoacoustically in the laboratory, using the electrical drive to a noise-generating loudspeaker as the reference. This system produced 10-20 dB of subjective noise-reduction between 250 Hz and 1 kHz, and smaller amounts at higher frequencies. The system was modified to operate in a real MR scanner where the reference was obtained by recording the acoustic scanner noise. Objective reduction by 30-40 dB of the most intense component in scanner noises was realized between 500 Hz and 3500 Hz, and subjective reduction of 12 dB and 5 dB in tests at frequencies of 600 Hz and at 1.9 kHz, respectively. Although the benefit of ANC is limited by transmission paths to the cochlea other than air-conduction routes from the auditory meatus, ANC achieves worthwhile attenuation even in the frequency range of maximum bone conduction (1.5-2 kHz). ANC should, therefore, be generally useful during auditory fMRI.     

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.     

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.     

Active noise control in finite length ducts

A simple technique for the active control of sound in ducts, initially suggested by Olson and May [1], is investigated in detail. A simple, “virtual earth” principle, feedback loop is used to drive the sound pressure to a minimum at a microphone placed close to a loudspeaker in the duct wall. This produces a reflection of downstream travelling plane waves. A detailed investigation of the loudspeaker near field has enabled the optimum position of the microphone to be identified. The system is shown to be especially effective at the frequencies of the longitudinal duct resonances, where the acoustic response of the duct produces a high loop gain. Results are presented which show a reduction of up to 20 dB in the amplitude of low frequency broadband noise at a position downstream of the cancelling source.     

Power requirements for active noise control in ducts

Active attenuation of noise in a duct generally requires either one or two rings of cancelling loudspeakers located around the duct perimeter. Consideration is given to the acoustic loading on the loudspeakers and it is shown that the use of a horn is likely to create more problems that it solves. Direct radiator operation, with the drive units attached directly to the duct walls, is preferable. The single ring (monopole) system reflects the noise giving rise to upstream standing waves, meaning that the loudspeakers and amplifiers must be able to handle correspondingly larger signals. The double ring (dipole) system absorbs the noise and is more efficient than the monopole system. The dipole system can be made still more efficient over a narrow band of frequencies by tuning both the loudspeakers and the spacing between them.     

Simulations and experiments for hybrid noise control systems

The purpose of this study is to explore the effects of sound elimination in a cylindrical duct by combining a reactive muffler and active noise control (ANC) system. Besides the exploration via experiment of the combined noise control system, a Grey prediction based on Grey theory is also applied to ANC for this hybrid system.In the experiment for this system, a combined adaptive algorithm is adopted. The results of sound elimination are compared between cases with ANC systems installed before the muffler and after the muffler. The results indicate that the sequence of arrangement of muffler can influence the results of active noise control. According to the results of experiment and simulation, the effect of noise reduction in ANC system is influenced extremely by reference signal received. The transmission loss and insertion loss in this system are also discussed in details. Besides, the experimental results indicate that the hybrid system has the advantages over a traditional muffler when the muffler is not designed for the frequency of the noise. Furthermore, the mathematic simulation for acoustic field in a cylindrical duct with a muffler is performed in order to verify the experiment results. Finally, Grey theory is applied to estimate the expected signals in order to perform a computer simulation of Grey prediction to explore effects of the ANC system. The results indicate that application of Grey theory gives a good control for the hybrid system.     

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.     

Side-branch resonators modelling with Green׳s function methods

This paper deals with strategies for computing efficiently the propagation of sound waves in ducts containing passive components. In many cases of practical interest, these components are acoustic cavities which are connected to the duct. Though standard Finite Element software could be used for the numerical prediction of sound transmission through such a system, the method is known to be extremely demanding, both in terms of data preparation and computation, especially in the mid-frequency range. To alleviate this, a numerical technique that exploits the benefit of the FEM and the BEM approach has been devised. First, a set of eigenmodes is computed in the cavity to produce a numerical impedance matrix connecting the pressure and the acoustic velocity on the duct wall interface. Then an integral representation for the acoustic pressure in the main duct is used. By choosing an appropriate Green?s function for the duct, the integration procedure is limited to the duct–cavity interface only. This allows an accurate computation of the scattering matrix of such an acoustic system with a numerical complexity that grows very mildly with the frequency. Typical applications involving Helmholtz and Herschel–Quincke resonators are presented.