基于波束形成缩放声强的声源局部声功率计算

基于波束形成法识别噪声源时,为计算主要噪声源的辐射声功率,给出了基于平面波模型的声强缩放方法,模拟计算了单极子点声源局部声功率的计算误差,结果显示:当阵列平面与声源计算平面间距离等于阵列直径时,基于波束形成缩放声强计算的声功率误差仅略高于0.1 dB。为克服旁瓣干扰,给出了具有一定动态范围的声源计算平面积分法,模拟计算了单极子点声源的局部声功率,结果表明:该积分法的计算值与主瓣区域积分法的计算值近似相等,均约等于理论声功率。进一步,波束形成法与声强法的对比算例试验验证了基于波束形成缩放声强计算声源局部声功率方法的有效性。      

Effect of room absorption on human vocal output in multitalker situations

People increase their vocal output in noisy environments. This is known as the Lombard effect. The aim of the present study was to measure the effect as a function of the absorption coefficient. The noise source was generated by using other talkers in the room. A-weighted sound levels were measured in a 108 m(3) test room. The number of talkers varied from one to four and the absorption coefficients from 0.12 to 0.64. A model was introduced based on the logarithmic sum of the level found in an anechoic room plus the increasing portion of noise levels up to 80 dB. Results show that the model fits the measurements when a maximum slope of 0.5 dB per 1.0 dB increase in background level is used. Hence Lombard slopes vary from 0.2 dBdB at 50 dB background level to 0.5 dBdB at 80 dB. In addition, both measurements and the model predict a decrease of 5.5 dB per doubling of absorbing area in a room when the number of talkers is constant. Sound pressure levels increase for a doubling of talkers from 3 dB for low densities to 6 dB for dense crowds. Finally, there was correspondence between the model estimation and previous measurements reported in the literature.     

Noise in the operating rooms of Greek hospitals

This study is an evaluation of the problem of noise pollution in operating rooms. The high sound pressure level of noise in the operating theatre has a negative impact on communication between operating room personnel. The research took place at nine Greek public hospitals with more than 400 beds. The objective evaluation consisted of sound pressure level measurements in terms of L(eq), as well as peak sound pressure levels in recordings during 43 surgeries in order to identify sources of noise. The subjective evaluation consisted of a questionnaire answered by 684 operating room personnel. The views of operating room personnel were studied using Pearson's X(2) Test and Fisher's Exact Test (SPSS Version 10.00), a t-test comparison was made of mean sound pressure levels, and the relationship of measurement duration and sound pressure level was examined using linear regression analysis (SPSS Version 13.00). The sound pressure levels of noise per operation and the sources of noise varied. The maximum measured level of noise during the main procedure of an operation was measured at L(eq)=71.9 dB(A), L(1)=84.7 dB(A), L(10)=76.2 dB(A), and L(99)=56.7 dB(A). The hospital building, machinery, tools, and people in the operating room were the main noise factors. In order to eliminate excess noise in the operating room it may be necessary to adopt a multidisciplinary approach. An improvement in environment (background noise levels), the implementation of effective standards, and the focusing of the surgical team on noise matters are considered necessary changes.     

线阵声强缩放方法在水下声源辐射功率估算中的应用

为了解决水下声源辐射声功率难以计算的问题,利用线阵声强缩放方法在波束形成声源识别的基础上,根据波束输出结果与声源辐射声功率之间的换算关系来获得相应的声功率。为了提高线阵声强缩放方法的水下声功率估算精度,给出了一定动态范围限制的主瓣区域积分方法,并通过仿真验证了该方法的有效性。在消声水池中开展了水下声功率估算的实验研究。在不同的测试距离下,对双声源条件下的单频以及宽带声源在阵列侧的辐射声功率进行了估算,以混响法的测量结果为参考值,研究了估算误差随声源频率、测试距离等影响因素的变化规律。实验结果表明,无论是单频还是宽带声源,声功率的最大估算误差不超过2.6 dB,在高频时不超过1.6 dB。验证了线阵声强缩放方法应用于水下声源辐射声功率估算的正确性与可行性。      

Verbal communication and noise in eating establishments

A new simple prediction model has been derived for the average A-weighted noise level due to many people speaking in a room with assumed diffuse sound field. Due to the feed-back influence of noise on the speech level (the Lombard effect), the speech level increases in noisy environments, and the suggested prediction model gives a 6 dB reduction of the noise level by doubling the equivalent absorption area of the room. This is in contrast to the lowering by 3 dB by doubling of the absorption area for a constant power sound source. The prediction model is verified by experimental data found in the literature. In order to achieve acceptable conditions for speech communication within a small group of people, a guide for the recommended minimum absorption area per person in eating establishments is provided.     

线阵波束形成声强缩放方法估算声源的辐射声功率

为了解决波束形成声源识别过程中声源辐射声功率定量计算的问题,给出了阵型简洁、便于组合的线阵声强缩放模型。通过推导线阵的声强缩放系数,建立起线阵波束输出结果与声源辐射声功率之间的换算关系。无论是线阵还是平面阵的声强缩放方法,对于偏离阵列中心位置较远处的声源进行辐射声功率估算时都存在较为明显的误差。通过理论推导和仿真模拟计算,研究了同一单极子点声源在不同位置处的声功率估算偏差随频率、幅度的变化规律,发现该估算偏差只与声源偏离位置有关,而与声源自身的强度信息无关的结论,据此给出了相应的声功率估算修正方法。半消声室实验结果和声压法测量结果对比表明:修正后的线阵声强缩放方法用于中高频声源的辐射声功率计算时,单频声源的估算误差不超过1.0 dB,宽带声源的估算误差不超过1.8 dB。      

Improved Statistically Optimal Nearfield Acoustical Holography in subsonically moving fluid medium

Statistically Optimal Nearfield Acoustical Holography (SONAH) can be used to reconstruct three-dimensional sound fields by projecting two-dimensional data measured on a “small” aperture that partially covers a composite sound source in a “static” fluid medium. Here, an improved SONAH procedure is proposed that includes the mean flow effects of a moving fluid medium while the sound source and receivers are stationary. The backward projection performance of the proposed procedure is further improved by using a wavenumber filter to suppress subsonic noise components. Through numerical simulations at Mach 0.6, it is shown that the improved procedure can accurately reconstruct sound source locations and radiation patterns: e.g., the spatially averaged reconstruction errors of the conventional and improved SONAH procedures are 15.40 dB and 0.19 dB, respectively, for a monopole simulation and 21.60 dB and 0.19 dB for an infinite-size panel. The wavenumber filter further reduces spatial noise, e.g., decreasing the reconstruction error from 1.73 dB to 0.19 dB for the panel simulation. An existing data measured in a wind tunnel operating at Mach 0.12 is reused for the validation. The locations and radiation patterns of the two loudspeakers are successfully identified from the sound fields reconstructed by using the proposed SONAH procedure.     

基于声压-质点速度声强探头的材料吸声系数的测量

通过由一个声压换能器和一个质点速度换能器所构成的传感器(p-u声强探头)同时测量材料表面附近的声压和质点振动速度,可直接得到其声学阻抗,进而得到材料的反射因子、吸声系数。本文利用一个p-u探头声强测量系统,在半消声室内测量了三聚氰胺泡沫的吸声系数,分析了声源高度和入射角度、材料样本尺寸和厚度对吸声系数测量的影响,并和阻抗管中测量得到的法向吸声系数进行了对比。最后分析了声阻抗率的幅值和相位误差对吸声系数的影响,推导了它们的误差传递公式。     

基于平面声源进行结构声辐射有源控制的实验研究

采用分布式平面声源作为次级声源,对振动钢板的声辐射进行了抵消实验,验证了以往研究中的一系列关键理论。实验研究结果表明:一个平面声源可以控制钢板奇-奇模态的声辐射,两个平面源可以控制结构偶-奇或奇-偶模态的声辐射,同时也可以控制结构奇-奇模态的声辐射;平面声源的面积和布放位置对降噪效果有重要影响,采用单个平面声源控制时,平面声源面积越大,控制效果越好;基于近场声压的误差传感策略是有效可行的,实际中,将近场测量面的声功率作为有源控制的目标函数与总声功率作为目标函数是一致的;控制后远场声压和声强都得到有效降低,部分区域的声能向声源流动,近场声压及声强分布也发生显著变化。     

Simple sound power measurements

A simple method of measuring sound power is described. Noise is added from a second calibrated source until the room sound pressure level is increased by 3 dB. The noise output of the device is then equal to the added power. A simple loudspeaker, modified to reduce the effect of the environment on the sound power output, is used as the calibrated source of sound power. Checks show that the sound power readings are substantially independent of the room in which they are made.     

Estimating the direct-to-reverberant energy ratio from the coherence between coincident pressure and particle velocity

An analytical expression for the relationship between the direct-to-reverberant energy ratio (DRR) and the coherence estimation function between coincident pressure and particle velocity component is derived. The analytical solution is first validated with simulated room impulse responses and then used to estimate the DRR in five octave bands for several receiver positions measured in a total of 11 rooms of vastly different sizes and acoustic characteristics. The accuracy is evaluated by comparison with the DRR estimated directly from the room impulse response. The difference is typically 5 dB. For two rooms, the variation of the DRR estimate with source-to-receiver position is also shown. The method is blind in the sense that it is virtually independent of the signal generated by a single sound source.     

Scan-based near-field acoustical holography and partial field decomposition in the presence of noise and source level variation

Practical holography measurements of composite sources are usually performed using a multireference cross-spectral approach, and the measured sound field must be decomposed into spatially coherent partial fields before holographic projection. The formulations by which the latter approach have been implemented have not taken explicit account of the effect of additive noise on the reference signals and so have strictly been limited to the case in which noise superimposed on the reference signals is negligible. Further, when the sound field is measured by scanning a subarray over a number of patches in sequence, the decomposed partial fields can suffer from corruption in the form of a spatially distributed error resulting from source level variation from scan-to-scan. In the present work, the effects of both noise included in the reference signals, and source level variation during a scan-based measurement, on partial field decomposition are described, and an integrated procedure for simultaneously suppressing the two effects is provided. Also, the relative performance of two partial field decomposition formulations is compared, and a strategy for obtaining the best results is described. The proposed procedure has been verified by using numerical simulations and has been applied to holographic measurements of a subsonic jet.     

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.     

椰胡声功率级的半消声室测试

椰胡是具有地方特色的拉弦乐器之一,多用于演奏广东音乐和广东戏曲、曲艺的伴奏。但对其发声强度一直未进行过科学的测试。本文介绍在华南理工大学半消声室内参照ISO(GB)标准对椰胡声功率级的测量工作。由两位资深乐师分别用两把椰胡在pp,mp,f和ff力度下演奏单音、音阶和乐曲,对每一把椰胡的每一测试内容,由十通道测试设备同步测试中心频率为100～10000 Hz的1/3倍频带声压级谱,通过计算获得每把椰胡在演奏每项内容时的声功率级和动态范围。通过对两把椰胡的测试结果进行平均,获得该乐器在演奏上述内容时的典型声功率级数值及频谱。文中并将半消声室内的测试结果与混响室内的测试结果相对比,探讨测试环境对测试结果的影响。民族乐器发声强度及其频谱特性的获得是开展民族音乐厅堂音质研究的基础。     

Prony's method applied to estimate noise source locations and their sound powers

An application of Prony's method for evaluating the acoustic power and location of sound sources from spatially sampled data is described. A sound source considered as a point source has an intensity proportional to the inverse square of the distance between source and observation point. The Fourier transform of this intensity function is an exponential function with a real exponent. The shift property of the Fourier transform results in a spectral change in the phase angle, which is expressed in the transform domain by a multiplicative exponential function of pure imaginary exponent. In this paper the usual time axis of the Fourier pair of time and frequency is treated as a variable denoting the location of the sound source. Accordingly, each spectral component of spatially sampled sound intensity generated by n point sources can be expressed as a linear combination of n complex exponentials. By applying Prony's method to the spectral data, these unknown exponents can be calculated numerically. This paper deals with an estimation procedure to find the location and power of a noise source. The estimation is done by minimizing the sum of the squares of the errors between the model and measured data. The proposed method has general applicability to problems where the so-called inverse square law for intensity can be assumed to be valid.     

Smart panel with active damping units. Implementation of decentralized control

This paper contains the second part of a study on a smart panel with five decentralized velocity feedback control units using proof mass electrodynamic actuators [Gonzalez Diaz et al., J. Acoust. Soc. Am. 124, 886 (2008)]. The implementation of five decentralized control loops is analyzed, both theoretically and experimentally. The stability properties of the five decentralized control units have been assessed with the generalized Nyquist criterion by plotting the loci of the eigenvalues of the fully populated matrix of frequency response functions between the five error signals and five input signals to the amplifiers driving the actuators. The control performance properties have been assessed in terms of the spatially averaged response of the panel measured with a scanning laser vibrometer and the total sound power radiated measured in an anechoic room. The two analyses have shown that reductions of up to 10 dB in both vibration response and sound radiation are measured at low audio frequencies, below about 250 Hz.     

Fundamentals of the direct measurement of sound intensity and practical applications

Sound intensity is a measure of the magnitude and direction of the flow of sound energy. Developments in sound intensity measurement capabilities in the last quarter century have occurred because of several reasons. The main ones include the derivation of the cross-spectral formulation for sound intensity and developments in digital signal processing. This paper begins with a brief historical introduction of sound intensity measurements. Then elementary theory for sound intensity is presented. A section on sound intensity measurements is then included. The next section of the paper discusses sources of measurement error; the major sources of error are described in some detail. The paper continues with a discussion of the main applications of sound intensity measurements: sound power determination, noise source identification, and transmission loss measurements. The paper concludes with a discussion of ISO and ANSI intensity related standards and relevant references.     

Enhancing maximum measurable sound reduction index using sound intensity method and strong receiving room absorption

The sound intensity method is usually recommended instead of the pressure method in the presence of strong flanking transmission. Especially when small and/or heavy specimens are tested, the flanking often causes problems in laboratories practicing only the pressure method. The purpose of this study was to determine experimentally the difference between the maximum sound reduction indices obtained by the intensity method, RI,max, and by the pressure method, Rmax. In addition, the influence of adding room absorption to the receiving room was studied. The experiments were carried out in an ordinary two-room test laboratory. The exact value of RI,max was estimated by applying a fitting equation to the measured data points. The fitting equation involved the dependence of the pressure-intensity indicator on measured acoustical parameters. In an empty receiving room, the difference between RI,max and Rmax was 4-15 dB, depending on frequency. When the average reverberation time was reduced from 3.5 to 0.6 s, the values of RI,max increased by 2-10 dB compared to the results in the empty room. Thus, it is possible to measure wall structures having 9-22 dB better sound reduction index using the intensity method than with the pressure method. This facilitates the measurements of small and/or heavy specimens in the presence of flanking. Moreover, when new laboratories are designed, the intensity method is an alternative to the pressure method which presupposes expensive isolation structures between the rooms.     

Comparison of the noise attenuation of three audiometric earphones, with additional data on masking near threshold

The noise-excluding properties of a standard supra-aural audiometric earphone, a widely used circumaural-supra-aural combination, and an insert earphone sealed to the ear with a vinyl foam eartip were measured in a diffuse-field room complying with ANSI S12.6-1984. Data on attenuation were obtained monaurally with the nontest ear plugged and muffed. Results for the supra-aural earphones generally agreed well with previously reported measurements. A broadband masking noise was used to directly test the ANSI S3.1-1977 permissible background noise levels for measuring to audiometric zero using standard audiometric earphones. This "ANSI noise" raised the average thresholds of 15 normal-hearing test subjects by 3 to 5 dB at the octave frequencies from 500 to 4000 Hz. With a noise conforming to the less stringent OSHA-1983 regulation, average thresholds were elevated 9 to 17 dB. An "ENT office noise" with an overall sound level of 54 dBA raised average thresholds even further, by as much as 29 dB at 500 Hz. Use of the circumaural system in the office noise limited the threshold elevation to 11, 5, 2, and 0 dB at the four octave frequencies tested. With the fully ("deeply") inserted foam eartips, the threshold elevation in the simulated office noise was 2 dB or less at all test frequencies. Actual threshold elevations agreed closely with predictions based on a critical ratio calculation utilizing measured sound field noise levels and measured earphone attenuation values.     

基于脉冲声分离的刚性球散射声控制

提出了一种基于脉冲声的三维空间中刚性球散射声分离方法,并利用前馈、固定系数控制方式对分离出的散射声进行有源控制,抑制散射声强度,实现了刚性球散射体在观测点处“声学不可见”。该方法利用脉冲信号作为初级噪声,通过有无刚性球时传声器采集脉冲信号的差值确定散射声大小,实现散射声与声源直达声的分离。对分离出的散射声进行多通道有源控制以验证该文所提分离方法及控制系统的有效性。实验结果表明,700~1000 Hz范围内,有源控制开启后,双通道散射声的平均降噪量大于5 dB,多通道散射声的平均降噪量大于8 dB,且误差传声器处采集的残余声场与无刚性球时采集的初级声场信号波形基本一致,实现了刚性球散射体在误差传声器处“声学不可见”。此外,参考传声器布放位置的选取问题也在该文做了详细讨论。