虚拟声屏障在变压器低频降噪中的实验研究

针对开口房间内的变压器,在开口处布放若干扬声器和误差传声器构成虚拟声屏障,实验研究了虚拟声屏障对通过开口向外辐射的低频线谱噪声的控制效果。将15个次级源近似均匀分布在面积为2 m′ 2.7 m的开口面上,左右间距约58.5 cm、上下间距45-65 cm,15个误差传声器分别位于对应的次级源正前方1 m,系统采用自适应谐波降噪算法。结果表明：虚拟声屏障系统在误差点100 Hz、200 Hz和300 Hz的平均降噪量分别达到12.7 dB、19.9 dB和22.2 dB,虚拟声屏障对100 Hz、200 Hz、300 Hz线谱噪声的控制效果与单层封闭窗户相当,且内部合成参考信号,无需外接参考传声器。采用虚拟声屏障对开口房间内的变压器降噪的好处是实现室内外的自然通风,便于变压器的散热。     

基于薄膜编码超表面的宽频超薄声散射体

该文提出了一种基于薄膜编码超表面的宽频超薄声散射体。利用附加质量块的薄膜和空气腔组成的薄膜结构构建了反射声波相位差接近180°的两种共振单元。将两种共振单元按照一定的顺序进行排列,可以组成深亚波长尺寸下的声学超表面。所构建的声学超表面可以产生宽频有效的散射声场。通过有限元仿真软件对多个频率的近场散射声场分布、远场声指向性和扩散系数进行了仿真计算,仿真结果显示,该散射体可以高效地散射入射声波,并且散射效果在一定的频率范围内是宽频有效的。     

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。     

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。     

充液管路系统流体声与结构声的复合有源控制

采用基于谐频自适应控制算法的有源消声与消振系统对充液管路系统突出的低频线谱噪声进行有源控制实验研究.建立了泵水循环管路实验系统,在管路中安装有源消声器对流体声进行控制,在管路出口障板上采用8×8通道有源消振系统控制结构声辐射。开展的低频线谱噪声与振动有源控制实验结果表明,在50~200 Hz频带内,通过结合有源噪声与振动控制可在多数频点取得10 dB以上的降噪效果。针对该实验系统,通过分别控制流体声和结构声分析了两者的贡献.实验结果验证了有源消声与消振系统具有较好的降噪性能,各频点处流体声与结构声占比情况不同,需要综合控制流体声与结构声才可以取得显著的降噪效果。      

外部加环肋有限长圆柱壳体声散射

为了深入理解双层加肋圆柱壳体的声散射机理,专门研究了两层壳体之间环肋的声散射。仅考虑模型的刚性散射,采用Kirchhoff近似推导了单个环肋的反向散射声场的解析解,并推广到等间距的环肋散射,结合圆柱壳刚性散射得到外部加周期性环肋的圆柱壳体的散射声场近似解。同时,利用图形声学方法(GRACO)对模型的目标强度进行数值计算。理论与实验的结果表明,刚性散射在反向散射声场中起主要作用,周期性环肋引起的Bragg散射对散射声场有重要贡献,同时遮挡效应在实际情况下有较大作用。      

有源头靠中一种分布式虚拟传声器优化方法

有源降噪头靠系统中,远程虚拟传声器技术能够解决控制点处与误差传声器处降噪量不匹配的问题。在实际应用中,多通道虚拟传声器技术存在收敛速度慢和运算复杂度高等问题。针对这个问题,本文通过重新设计远程虚拟传声技术的离线优化过程,提出一种分布式远程虚拟传声器技术优化方法。该方法将虚拟次级通路矩阵作对角化限制,同时对观测传递函数矩阵进行联合寻优,以实现一种分布式的更新算法。有源降噪头靠实验结果表明,所提算法能够有效降低远程虚拟传声器技术算法的运算复杂度,并且提升了算法的收敛速度。     

采用分布式球形阵列的球谐波域声场分离方法

为从测得的混合声场信号中提取出需要的目标声场,提出一种基于分布式球形传声器阵列的声场分离方法。该方法依据声场的球谐波分解,利用阵列各传声器采集到的声场声压信号,获得目标声场与干扰声场的球谐波展开系数,进而估计目标声场。该方法利用声场以不同中心展开的球谐波系数之间的变换关系,直接建立传声器测量声压信号与整体坐标系下声场展开系数的方程,与传统的分布式球形阵列声场分离方法,即先求解局部坐标系下声场展开系数,再变换为整体展开系数的方法,进行比较。分别通过数值仿真和实验说明了提出方法的有效性。结果表明:该方法能够从混合声场中较准确地估计出目标声场,并且在干扰声场能量增大时,保持了较小的声场估计误差,相比于传统方法误差增加更少。     

水下流激孔腔噪声前馈有源控制实验研究

水流流经腔体时引起的自激振荡会形成流激噪声,该文通过仿真和实验研究了水下流激孔腔噪声的主动声场控制。应用有源前馈控制方法,对比了参考信号的选取以及控制器中增加的泄露因子对控制结果的影响,并在水循环管路中对流激孔腔噪声前馈控制效果进行了实验验证。研究结果表明,前馈有源噪声控制方法可有效抑制水下流激孔腔噪声,在误差点和辅助观测点都取得了良好的降噪效果,最大降噪量大于8 dB,并且使用腔内信号作为参考的控制效果优于上游信号作为参考。     

一种基于虚拟传感的无需误差传声器的自适应有源降噪方法*

该文基于虚拟传感技术引入了一种用于耳机的无需误差传声器的自适应有源降噪方法。该算法仅使用一个参考传声器实现了一种前馈和反馈自适应算法结合的有源降噪算法,提高了有源降噪稳定性,简化了耳机硬件结构。利用DSP平台实现了该文提出的方案,并通过实验验证了其良好的降噪性能和实用价值。     

Directional cancellation of acoustic noise for home window applications

This paper focuses on an active noise cancellation system for a home window using a transparent acoustic transducer. In a traditional active noise cancellation system, direct microphone measurements are used for reference and error signals. In the case of the window application, both external and internal sound would be picked up by such microphones. This leads to adverse effects on the performance of the active noise cancellation system and also to distortion of the internal sound. To address this problem, a wave separation technique is proposed to separate the internal and external components of sound. The wave separation algorithm is based on the use of two microphones and an algorithm that separates components based on their direction of travel. An active noise cancellation system is implemented using wave separation for both the error and reference signal measurements. The performance of the resulting ANC system is experimentally tested in a cabin equipped with a window and results are presented. Experimental results show that the new system is able to accurately preserve desired internal sound while cancelling uncorrelated external noise.     

单层传声器阵列信号空间重采样的声波分离方法

为了重构非自由声场中目标声源的声场响应,提出单层传声器阵列信号空间重采样的声波分离方法.以球面波函数为基函数,建立由系列球面波函数叠加表达的声场数学模型.基于近场声全息原理,利用单层传声器阵列面上空间重采样形成的两组声压测量信号,求解基函数系数,并重构出传声器阵列两侧声源各自的声场响应,实现声波分离.使用脉动球和振动球共同作用的非自由声场,检验了数学模型以及传声器信号信噪比、传声器阵列形状和面积、声源中心位置、频率等关键参数对声波分离精度的影响,并在全消声室内进行了实验验证.最后,对单层传声器阵列重采样的声波分离方法的实施给出了建议.     

Improving robustness of filtered-x least mean p-power algorithm for active attenuation of standard symmetric-α-stable impulsive noise

The paper concerns active control of impulsive noise having peaky distribution with heavy tail. Such impulsive noise can be modeled using non-Gaussian stable process for which second order moments do not exist. The most famous filtered-x least mean square (FxLMS) algorithm for active noise control (ANC) systems is based on the minimization of variance (second order moment) of error signal, and hence, becomes unstable for the impulsive noise. In order to improve the robustness of adaptive algorithms for processes having distributions with heavy tails (i.e. signals with outliers), either (1) a robust optimization criterion may be used to derive the adaptive algorithm or (2) the large amplitude samples may be ignored or replaced by an appropriate threshold value. Among the existing algorithms for ANC of impulsive noise, one is based on the minimizing least mean p-power (LMP) of the error signal, resulting in FxLMP algorithm (approach 1). The other is based on modifying; on the basis of statistical properties; the reference signal in the update equation of the FxLMS algorithm (approach 2). In this paper we propose two solutions to improve the robustness of the FxLMP algorithm. In first proposed algorithm, the reference and the error signals are thresholded before being used in the update equation of FxLMP algorithm. As another solution to improve the performance of FxLMP algorithm, a modified normalized step size is proposed. The computer simulations are carried out, which demonstrate the effectiveness of the proposed algorithms.     

Incorporation of loudness measures in active noise control

An attempt has been made to use a modified version of a standard active noise control algorithm in order to take into account the unique response of the human auditory system. It has been shown in the past that decreasing the sound pressure level at a location does not guarantee a similar decrease in the perceived loudness at that location. Typically, active noise control is based on minimizing the "error signal" from a mechanical device such as a microphone, whose response is nominally flat across the frequency response range of the human ear. However, if the response of the ear can be approximated by digitally filtering the error signal before it reaches the adaptive controller, one can, in effect, minimize the more subjective loudness level, as opposed to the sound pressure level. The work reported here entails simulating active noise control based upon minimizing perceived loudness for a collection of input noise signals. A comparison of the loudness of the resulting error signal is made to the loudness of that resulting from standard sound pressure level minimization. It has been found that the effectiveness of this technique is largely dependent upon the nature of the input noise signal. Furthermore, this technique is judged to be worth considering for use with applications of active noise control where the uncontrolled noise more prominently constitutes low range audio frequencies (approximately 30 Hz-100 Hz) than medium range audio frequencies (approximately 300 Hz-600 Hz).     

Determination of the sound energy level of a gunshot and its applications in room acoustics

In some cases an impulsive noise source such as a gunshot can be a preferred alternative when investigating building acoustics, including sound insulation measurements, when compared to conventional steady state noise sources. A gun equipped with blank cartridges is an impulsive noise source that is lightweight and small enough to be easily transported. The differences in the noise characteristics between individual cartridges for the same gun are usually small, so the impulsive source can be replicated to a high degree. This paper is focused on the practical application of the sound exposure levels produced by a gunshot with a known sound energy level in the rooms under investigation. In this way, the equipment and methods required by the conventional method are simplified significantly. Furthermore, reverberation times need not be measured, since the equivalent absorption area can be directly obtained from the measured sound exposure levels. Using Green’s theorem, the roles of the sound source and measuring microphone were exchanged, which simplified the determination of sound insulation as it was easier to change the position of the gun than the microphone. The results obtained using the impulsive noise source were in good agreement with those obtained using the conventional method. Above 100 Hz, their difference in any frequency band of interest was less than 1 dB.     

Comparison of methods for processing acoustic intensity from orthogonal multimicrophone probes

One design for three-dimensional multimicrophone probes is the four-microphone orthogonal design consisting of one microphone at an origin position with the other three microphones equally spaced along the three coordinate axes. Several distinct processing methods have been suggested for the estimation of active acoustic intensity with the orthogonal probe; however, the relative merits of each method have not been thoroughly studied. This comparative study is an investigation of the errors associated with each method. Considered are orthogonal probes consisting of matched point sensor microphones both freely suspended and embedded on the surface of a rigid sphere. Results are given for propagating plane-wave fields for all angles of incidence. It is shown that the lowest error for intensity magnitude results from having the microphones in a sphere and using just one microphone for the pressure estimate. For intensity direction, the lowest error results from having the microphones in a sphere and using Taylor approximations to estimate the particle velocity and pressure.     

Active control of radiation from a piston set in a rigid sphere

Active control of the sound radiated from a piston set in a rigid sphere with a set of control point sources around is considered in this paper, where the scattering sound field of the control sound from the rigid sphere has been taken into account to minimize the total radiated sound power. Analytic results of the sound power are obtained and numerical simulations show that it is possible to reduce the radiation from a small piston set in a rigid sphere similar to the size of a human head up to a certain frequency. It is found that the introduction of the scattering object makes significant differences from the active control without scattering objects. This being the case, the scattering object makes the active noise control easier. To increase the global reduction of sound-power output, the optimal number and locations of the control sources and the optimal number and locations of error sensors are discussed. Finally, experiments with one control source and one error sensor around a head simulator have been carried out to verify the simulation results.     

Experimental results for multichannel feedforward ANC with noninvasive system identification

This paper presents experimental validation of a class of algorithms designed to enable active noise control (ANC) to function in environments when transfer functions change significantly over time. The experimental results presented are for broadband, local quieting in a diffuse field using a multichannel ANC system. The reverberant enclosure is an ordinary room, measuring approximately 1.4 x 2.4 x 2.4 m3 and containing a seated occupant, with six microphones defining the quiet zone near the occupant's ears. The control system uses a single reference signal and two error channels to drive four secondary sources. Using an ideal reference sensor, reduction in sound pressure level is obtained at the quiet-zone microphones averaged over the frequency range 50 to 1000 Hz with an occupant seated in the room. Two main results are presented: first for an adaptive cancelling algorithm that uses static system models, and second for the same algorithm joined with a noninvasive real-time system identification algorithm. In the first case better than 23 dB of performance is obtained if the occupant remains still through calibration and testing. In the second case, approximately 18 dB is obtained at the error microphones regardless of the motion of the occupant.     

Hybrid active and passive control of fan noise

This paper deals with the global reduction of axial flow fan noise in ducts in a building using a hybrid passive-active noise control method. The effectiveness of using an infra-red device as a reference signal source is also investigated. It is shown that using such a hybrid noise control system over an axial-flow fan reduces the overall sound pressure level by 5 dB(A) in the surrounding environment and global control of the blade passing frequency can also be achieved. This paper also shows that using an infra-red device as a reference signal source produces marginally better control as compared with using a microphone reference sensor. Moreover, long term stability is guaranteed and the possibility of acoustic feedback is eliminated.