A note on sound radiation from distributed sources

The power output from a normally vibrating strip radiator is expressed in alternative general forms, one of these being chosen to refine and correct some particular estimates given by Heckl for different numerical ratios of strip width to wave length. An exact and explicit calculation is effected for sinusoidal velocity profiles when the strip width equals an integer number of half wave lengths.     

Distance rendering and perception of nearby virtual sound sources with a near-field filter model

Headphone rendering of nearby virtual sound sources represents to date an open issue in 3-D audio, due to a number of technical challenges and temporal requirements involved in the measurement of individual Head-Related Transfer Functions (HRTFs). In order to tackle this problem, we propose a filter model of near-field effects based on the Distance Variation Function (Kan et al., 2009). Thanks to its simple structure and low order, the model can be applied to any far-field virtual auditory display to yield a realistic and computationally efficient near-field compensation of spectral and binaural effects. The model is subjectively evaluated in two psychophysical experiments where the relative distance of pairs of virtually rendered sound sources is judged. Results show that even though sound intensity overshadows subtler near-field effects when it is available as a cue for distance, the model is capable of offering relative distance information of near lateral virtual sources when intensity cues are removed. Furthermore, performances of the model in relative distance rendering are compared to those of alternative near-field rendering methods available in the literature.     

Characterisation of valves as sound sources: Structure-borne sound

Taps and other valves are major sound sources in piping systems and can cause unacceptable noise levels in buildings. The noise results from the fluid-, structure- and air-borne sound emission. At present the acoustic emission of water appliances is tested according to a European standard, the shortcomings of which are apparent as a result of a round robin test of different European laboratories. As a result, there are currently neither acceptable measurement methods for water appliances available nor input data for prediction models. This paper considers methods of characterizing water appliances as sources of structure-borne sound. The concepts of mobility and free velocity are employed for a source characterisation based on power. Taps are considered alone and also in combination with a basin, where again the mobility and free velocity are used. A reception plate method is assessed as an alternative. The two methods each provide an independent characterisation of a structure-borne sound source as a single value. The values are on a power basis and provide input data suitable for prediction of the installed structure-borne power and thence the resultant sound pressure in adjacent rooms. Measured and predicted values of sound pressure level, caused by a wash-basin installed in an adjacent room, are compared.     

Characterisation of valves as sound sources: Fluid-borne sound

The sound pressure level in receiving rooms, caused by taps at the ends of pipe systems, is considered. The structure-borne sound power, from the pipes to the supporting wall, was obtained from intensity measurement of the fluid-borne sound power of the tap. The fluid-borne sound power is combined with a ratio of structure-borne sound power to fluid-borne sound power, obtained from laboratory measurements of similar pipe assemblies. Alternatively, a reception plate method is proposed, which avoids the necessity for intensity measurements. The structure-borne power into walls, to which the pipe work is attached, provides input to the standard building propagation model, which yields the predicted sound pressure level in the adjacent room.     

Auditory perception of sound source velocity

In this study we investigate the perception of the velocity of linearly moving sound sources passing in front of a listener. The binaural simulation of motion used in two psychoacoustical experiments includes changes in the overall sound pressure level, the Doppler effect, and changes in interaural time differences. These changes are considered as cues for the perception of velocity. The present experiments are an extension of the experiments performed by Lutfi and Wang [J. Acoust. Soc. Am. 106, 919-928 (1999)]. The results of Experiment I show that the differential velocity threshold is independent of the reference velocity (10, 20, 30, and 40 m/s), varying across listeners from 1.5 to 4.6 m/s. In Experiment II, a method based on the successive elimination of cues in compared pairs of signals was employed to estimate the weights of potential cues for velocity discrimination. The magnitudes of all underlying cues at thresholds are reported. The experimental results show the subject's preference for the Doppler cue and a weakest sensitivity to the cue related with interaural time differences. Finally, it was found that spatial differences in the source location at the endpoints of the motion trajectory are not a significant factor in the velocity discrimination task.     

Identifying turbulent jet sound sources

Theoretical models of the formation of the acoustic field of a nonisothermal turbulent flow are analyzed. The paper presents possible methods of constructing solutions to the convective wave equation for a shear parallel nonisothermal turbulent flow. A simplified mathematical model of noise sources in the field of a nonisothermal shear turbulent flow is considered, based on real physical notions on the processes occurring in the jet mixing range, making it possible to approximately estimate the spectral noise characteristics.     

双声源位置和强度识别的三维声强方法

目前在远场识别声源空间位置和强度缺乏行之有效的方法。针对此问题,提出采用四传声器进行三维声强测量,从而构建出声强、声源坐标和声功率的非车线性方程组,求解方程得出声源空间坐标和强度的方法。以3个三维声强探头对两个同频率单极子声源的识别为例,分别利用数值仿真和半消声室内的实验进行方法验证,并对声源的识别空间分辨率做了测试,得出角度识别最大误差为3.83°,为真实值的8.5%,距离识别最大误差0.1 m,为真实距离的10%。结果表明采用该方法空间坐标和声功率识别均具有很高的准确度,双声源的空间位置分辨力也优于远场声全息方法。     

Imaging of directional distributed noise sources

This study relates to the acoustic imaging of noise sources that are distributed and strongly directional, such as in turbulent jets. The goal is to generate high-resolution noise source maps with self-consistency, i.e., their integration over the extent of the noise source region gives the far-field pressure auto-spectrum for a particular emission direction. Self-consistency is possible by including a directivity factor in the formulation of the source cross-spectral density. The resulting source distribution is based on the complex coherence, rather than the cross-spectrum, of the measured acoustic field. For jet noise, whose spectral nature changes with emission angle, it is necessary to conduct the measurements with a narrow-aperture array. Three coherence-based imaging methods were applied to a Mach 0.9 turbulent jet: delay-and-sum beamforming; deconvolution of the beamformer output; and direct spectral estimation that relies on minimizing the difference between the measured and modeled coherences of the acoustic field. The delay-and-sum beamforming generates noise source maps with strong spatial distortions and sidelobes. Deconvolution leads to a five-fold improvement in spatial resolution and significantly reduces the intensity of the sidelobes. The direct spectral estimation produces maps very similar to those obtained by deconvolution. The coherence-based noise source maps, obtained by deconvolution or direct spectral estimation, are similar at small and large observation angles relative to the jet axis.     

Blind extraction and localization of sound sources using point sources based approaches

This paper presents theoretical models for blind sound source localization and separation of the signals emitted by arbitrary point sources in free space. Source localizations are achieved by a model based approach that accounts for the spherical spreading of an acoustic wave and utilizes an iterative triangulation, based on the signals measured by a three-dimensional microphone array. Once source locations are determined, the source signals are separated by using the point source separation (PSS) method, which is valid for all types of signals, including harmonic, continuous, transient, random, narrowband and broadband. General solutions for signals separation are presented. Theoretically, PSS can reconstruct the individual source signals exactly. This is because it employs the free-space Green's function, which defines the exact correlation among individual sources and measurement microphones. To validate PSS, numerical simulations are carried out and results are compared with those obtained by FastICA (Independent Component Analysis) code. The impacts of various parameters such as the microphone configuration, type of source signals, signal to noise ratio, number of microphones and source localization errors on the quality of signals separation by using PSS and FastICA are examined. The advantages and disadvantages of PSS and FastICA are compared and discussed.     

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。     

A note on moving multipole sources of sound

The object of this note is to draw attention to the need, when speaking of a moving multipole source of sound, to state whether it is based on the wave equation for the velocity potential or the wave equation for the perturbation pressure. The theory of moving multipole sources of sound is firmly established, but, owing to lack of precision, some seemingly contradictory results are to be found in the literature. The reasons for these seemingly contradictory results are examined, and various basic relationships between a moving multipole sound field and its method of generation are established.     

基于平面声源实施结构声辐射有源控制的理论研究

摘要研究了利用分布式平面声源对结构声辐射进行有源控制的问题。首先建立了系统的数学模型,然后推导了有源控制条件下次级声源的强度和声功率降低的计算公式。在实际应用中,次级声源参数(面积大小、安放位置、个数等)对控制效果有重要影响,本文基于有源控制的物理机理和数值仿真研究这些问题。结果表明: -般情况下,次级声源板的振动模态分布与初级结构振动模态分布不相同,因此,在低频范围内,需要至少4个分布式次级声源,方能有效地控制初级结构声辐射。     

Localization of multiple sound sources with two microphones

This paper presents a two-microphone technique for localization of multiple sound sources. Its fundamental structure is adopted from a binaural signal-processing scheme employed in biological systems for the localization of sources using interaural time differences (ITD). The two input signals are transformed to the frequency domain and analyzed for coincidences along left/right-channel delay-line pairs. The coincidence information is enhanced by a nonlinear operation followed by a temporal integration. The azimuths of the sound sources are estimated by integrating the coincidence locations across the broadband of frequencies in speech signals (the "direct" method). Further improvement is achieved by using a novel "stencil" filter pattern recognition procedure. This includes coincidences due to phase delays of greater than 2pi, which are generally regarded as ambiguous information. It is demonstrated that the stencil method can greatly enhance localization of lateral sources over the direct method. Also discussed and analyzed are two limitations involved in both methods, namely missed and artifactual sound sources. Anechoic chamber tests as well as computer simulation experiments showed that the signal-processing system generally worked well in detecting the spatial azimuths of four or six simultaneously competing sound sources.     

Optimisation of ground attenuation for moving sound sources

In an earlier study, Attenborough and Li [Attenborough K, Li, KM. Ground effect for A-weighted noise in the presence of turbulence and refraction. J Acoust Soc Am 102:1997;1013-22] derived a closed form analytical formula to calculate optimum ground parameters for reducing the A-weighted noise due to a stationary point source. This paper extends this earlier study by deriving an expression to calculate the sound field due to a source moving at a constant speed above a ground surface. An A-weighted mean-square sound pressure has been derived that enables one to estimate the sound exposure levels of a moving source. Numerical calculations for a realistic range of speeds show the influence of the source motion on the noise levels. Although the predicted effects on the ground attenuation are moderate, they are significant at a relatively low Mach number. In addition, the sensitivity to the atmospheric turbulence and the geometrical parameters tends to be altered by the source motion. On the other hand, it is demonstrated that the optimum ground parameters are similar for stationary and moving sources.     

Three-dimensional sound source localization using distributed microphone arrays

To improve the performance of sound source localization based on distributed microphone arrays in noisy and reverberant environments,a sound source localization method was proposed.This method exploited the inherent spatial sparsity to convert the localization problem into a sparse recovery problem based on the compressive sensing(CS) theory.In this method two-step discrete cosine transform(DCT)-based feature extraction was utilized to cover both short-time and long-time properties of the signal and reduce the dimensions of the sparse model.Moreover,an online dictionary learning(DL) method was used to dynamically adjust the dictionary for matching the changes of audio signals,and then the sparse solution could better represent location estimations.In addition,we proposed an improved approximate l_0norm minimization algorithm to enhance reconstruction performance for sparse signals in low signal-noise ratio(SNR).The effectiveness of the proposed scheme is demonstrated by simulation results where the locations of multiple sources can be obtained in the noisy and reverberant conditions.     

一种利用分布式传声器阵列的声源三维定位方法

为了提高噪声和混响条件下分布式传声器阵列进行声源定位的性能,提出一种利用空间稀疏性和压缩感知原理的声源三维定位方法。该方法首先通过两次离散余弦变换方式提取出声音信号特征,并用该特征来构建稀疏定位模型,以便能够综合利用语音信号的短时和长时特性,同时降低模型维数;然后利用在线字典学习技术动态调整字典,克服稀疏模型与实际信号之间的失配问题,增强稀疏定位模型的鲁棒性;进而提出一种改进的平滑l₀范数稀疏重构算法来进行声源位置解算,以提高低信噪比条件下的重构精度。仿真结果表明该方法不仅可以实现多目标定位,而且具有较强的抗噪声和抗混响能力.     

Sound-producing sources as objects of perception: rate normalization and nonspeech perception

In a variety of experiments and paradigms, researchers have attempted to determine whether or not speech perception is specialized by comparing perception of speech syllables to perception of nonspeech analogs. While nonspeech analogs appear optimal as comparisons to speech because they are acoustically similar without being recognized as speechlike, it is argued that the comparison they offer is confounded and uninterpretable. Two experiments are designed to show that, in auditory perception generally where acoustic signals are causal consequences of mechanical events, perceptual experiences are of the mechanical events themselves, not of the acoustic signal. This has two consequences. One is that there is a confounding in comparisons of speech with sine wave analogs that, whereas the one perceived as speech also has a definite causal source, the other, perceived as nonspeech, has an indeterminate or ambiguous source. A second is that response patterns in classification tasks such as those used in the literature comparing speech to nonspeech will reflect properties of the perceived sound-producing event; they will not provide a clear window on auditory system processes used to recover event properties. Experiment 3 is designed to show that perception of many acoustic-signal-producing events can appear to be special by the logic of speech-sine wave comparisons--even events that cannot plausibly be supposed to involve a specialization.     

基于时间反转的复杂声场拾声传声器阵列性能研究

探讨时间反转技术在复杂声场传声器阵列拾声中应用的可行性及其机理,给出其一般规律和性能。研究表明:在自由空间中,其拾声性能与频率,阵列形状和半径有关,频率越高,半径越大,拾声效果越好。在普通房间中,在语音频段内,圆弧阵列在预定目标点处的阵列增益性能要比离预定目标点约25 cm远处的位置处大5 dB以上。在普通房间和混响室中的实验验证了上述结论。