Separation of zeros for source signature identification under reverberant path conditions

This paper presents an approach to distinguishing the zeros representing a sound source from those representing the transfer function on the basis of Lyon's residue-sign model. In machinery noise diagnostics, the source signature must be separated from observation records under reverberant path conditions. In numerical examples and an experimental piano-string vibration analysis, the modal responses could be synthesized by using clustered line-spectrum modeling. The modeling error represented the source signature subject to the source characteristics being given by a finite impulse response. The modeling error can be interpreted as a remainder function necessary for the zeros representing the source signature.     

Improving syllable identification by a preprocessing method reducing overlap-masking in reverberant environments

Overlap-masking degrades speech intelligibility in reverberation [R. H. Bolt and A. D. MacDonald, J. Acoust. Soc. Am. 21(6), 577-580 (1949)]. To reduce the effect of this degradation, steady-state suppression has been proposed as a preprocessing technique [Arai et al., Proc. Autumn Meet. Acoust. Soc. Jpn., 2001; Acoust. Sci. Tech. 23(8), 229-232 (2002)]. This technique automatically suppresses steady-state portions of speech that have more energy but are less crucial for speech perception. The present paper explores the effect of steady-state suppression on syllable identification preceded by /a/ under various reverberant conditions. In each of two perception experiments, stimuli were presented to 22 subjects with normal hearing. The stimuli consisted of mono-syllables in a carrier phrase with and without steady-state suppression and were presented under different reverberant conditions using artificial impulse responses. The results indicate that steady-state suppression statistically improves consonant identification for reverberation times of 0.7 to 1.2 s. Analysis of confusion matrices shows that identification of voiced consonants, stop and nasal consonants, and bilabial, alveolar, and velar consonants were especially improved by steady-state suppression. The steady-state suppression is demonstrated to be an effective preprocessing method for improving syllable identification by reducing the effect of overlap-masking under specific reverberant conditions.     

混响场中用相关传声器识别声源直达声的近似工程法

本文研究了在混响场中用相关传声器识别主声源直达声的近似工程法，包括理论及实验技术。采用该方法对一个置于混响声污染下的恒定声源进行了直达声测量，并将测量结果与该恒定声源的消声室测量结果进行了比较，结果表明：相关传声器近似工程法在很大程度上能减少混响声对主要源声压测量的干扰，从而能在混响声场中获取一定精度的主声源声压级。     

非消声水池中低频换能器测量的空间域处理方法

提出了一种在非消声水池中低频换能器测量的空间域处理方法。该方法是在发射和接收换能器保持相对位置不变的情况下,通过多次无规则变换其在有限水域中的位置,让同频反射声的干扰贡献随机和无规则化,最终将上述所有位置接收的信号同步叠加而将同频反射声干扰消除,从而提取出了所需的直达波波形信号,克服了传统的脉冲测量技术受同频反射声干扰无法提取直达信号的问题。在5 m×7 m×6 m非消声水池,测量换能器样品,测量频率范围是400~2500 Hz,测量得到的结果与湖上测试结果相比,两者基本一致。该方法的实现,大大的提高了非消声水池的低频换能器测试能力,有效测量频率下限比一般脉冲测试方法降低一个数量级。     

Simulating the response of multiple microphones to a single acoustic source in a reverberant room

A method is presented for simulating the impulse response between an acoustic source and multiple microphones in a reverberant room. The method is similar to the image method described by Allen and Berkley [J. Acoust. Soc. Am. 65, 943-950 (1979)] but includes modifications to simulate received echo arrival time accurately. The essential modification is to represent each received echo as a low-pass-filtered impulse at the correct arrival time. Using this "low-pass impulse" method, reverberant rooms can be simulated with sufficient accuracy to investigate multiple-microphone systems that are sensitive to interchannel phase.     

A capacitive method for hall effect measurements

A new, simple and reliable method for Hall effect measurements is introduced. The method employs a capacitive probe technique and requires neither special shaping of samples nor probing Hall contacts. With this method, Hall effect measurements onp-Ge have been first extended to high electric fields up to 3kV/cm at 4.2 K. The wide applicability of this method is discussed.     

音乐厅响度评价新指标

响度是评价厅堂音质的最基本的参量之一,然而由于尚缺乏一个公认和可信的评价指标,如何评价厅堂响度的问题至今仍未妥善解决。本文建议将乐队齐奏强音标志乐段的平均声压级L_pF作为描述厅堂响度的指标,并给出了预计大厅L_pF值分布的计算方法。本文还示出两个厅堂L_pF的实测值并将之与预计算值作了比较,证实本计算方法的可行性。本文最后讨论了L_pF的优选值和容许值。     

A study on the random array generation method for moving sound source identification

Based on the problem that the generating method of random array structure is inefficient,a method is proposed to generate the random target arrays by using coaxial circular array in the polar coordinates in the premise that the array angular resolution of source identification is guaranteed.According to the principle of moving sound source identification,this work deduces the basic non-equidistance coaxial circular rings'radius,and generates target random arrays which were suitable for moving sound source identification through array partitioning,condition filtering in the polar coordinates and simulation evaluation.Finally,numerical simulation and moving car sound source identification test have been done.The analytical results show that using this method to generate random array is effective.Compared with the traditional regular arrays,the target random array has more accurate moving sound source identification performance.     

Optimizating the seats surrounding the platform in a recital hall when considering source directivity

The influence of source directivity can be significant for solo and chamber music performances, particularly for arena type halls that have improved visual intimacy for the audience. Computer simulation was performed to analyze the possibility of optimizing the acoustical qualities of a 4000-m³ recital hall with a significant portion of seats surrounding the platform when considering source directivity. The effects of surface treatments, length-to-width proportion, seating arrangement and platform location were evaluated. Most acoustical parameters derived from a baritone singer source were categorized as one factor that can be best represented by 2-kHz band early sound strength G_E. Regardless of room proportion, front-back 2-kHz band energy difference derived from optimized models can be controlled within 4 dB, a value much smaller than the difference in direct component.     

Effect of source spectrum on sound localization in an everyday reverberant room

Two experiments explored how frequency content impacts sound localization for sounds containing reverberant energy. Virtual sound sources from thirteen lateral angles and four distances were simulated in the frontal horizontal plane using binaural room impulse responses measured in an everyday office. Experiment 1 compared localization judgments for one-octave-wide noise centered at either 750 Hz (low) or 6000 Hz (high). For both band-limited noises, perceived lateral angle varied monotonically with source angle. For frontal sources, perceived locations were similar for low- and high-frequency noise; however, for lateral sources, localization was less accurate for low-frequency noise than for high-frequency noise. With increasing source distance, judgments of both noises became more biased toward the median plane, an effect that was greater for low-frequency noise than for high-frequency noise. In Experiment 2, simultaneous presentation of low- and high-frequency noises yielded performance that was less accurate than that for high-frequency noise, but equal to or better than for low-frequency noise. Results suggest that listeners perceptually weight low-frequency information heavily, even in reverberant conditions where high-frequency stimuli are localized more accurately. These findings show that listeners do not always optimally adjust how localization cues are integrated over frequency in reverberant settings.     

识别运动声源随机阵列生成法研究

针对随机阵列结构设计方面欠缺高效的生成方法这一问题,在保证阵列对声源识别精度的前提下,提出一种在极坐标下用于识别运动声源的随机阵列生成方法。根据声阵列识别运动声源的原理,推导了非等间距基本同轴圆环的半径,再通过降列分区、极坐标下条件筛选和模拟评价三个步骤,生成适用于识别运动声源的目标随机阵列,最后进行数值模拟和运动汽车噪声源识别实验进行验证。研究结果表明,用该方法能够高效地生成目标随机阵列,与常用规则阵列相比具有更良好的声源识别特性,并且具有准确的运动声源识别性能。     

A new mechanism for the anomalous hall coefficient with application to liquid metals

The effect of spin-orbit coupling of conduction electrons on the Hall coefficient of liquid metals is studied and a new mechanism for the anomalous Hall coefficient is suggested. S.O. coupling causes asymmetric scattering and a current, perpendicular to the magnetic field and to the electric field, is set up. This leads to an additional Hall coefficient , as was first calculated by Luttinger. But S.O. coupling also causes a dipole to appear near each ion, leading to an electrical polarization and an additional Hall coefficient,R _so.R _so is calculated for liquid normal metals and transition metals to first order in S.O. coupling. In normal metals,R _so and are proportional to the Pauli susceptibility. Both are positive for electron-like charge carriers. S.O. coupling leads to a maximum deviation from nearly free electron value in the Hall coefficient of a heavy, four or five-valent liquid metal with short mean free path. In transition metals, different expressions are obtained forR _so depending whether localized moments are present (magnetic model) or not (non-magnetic model). The sign ofR _so is determined by the value of thed-phase shift alone.R _so is proportional to the susceptibility ofd-electrons and can be large, leading to a positive Hall coefficient in the liquid transition metal.     

识别运动声源随机阵列生成法研究

针对随机阵列结构设计方面欠缺高效的生成方法这一问题,在保证阵列对声源识别精度的前提下,提出一种在极坐标下用于识别运动声源的随机阵列生成方法。根据声阵列识别运动声源的原理,推导了非等间距基本同轴圆环的半径,再通过降列分区、极坐标下条件筛选和模拟评价三个步骤,生成适用于识别运动声源的目标随机阵列,最后进行数值模拟和运动汽车噪声源识别实验进行验证。研究结果表明,用该方法能够高效地生成目标随机阵列,与常用规则阵列相比具有更良好的声源识别特性,并且具有准确的运动声源识别性能。     

基于近场声全息理论的运动声源动态识别方法

运动声源的准确识别是对其进行有效控制的重要前提,本文在近场声全息理论基础上,利用时域多普勒效应消除原理、球面波传播原理,建立了用于运动噪声源识别的动态识别方法。模拟计算和实车试验证明该识别方法能对高速运动噪声源进行准确识别。     

三维障碍物形状反演问题的一种简单算法

研究了从声散射场的远场分布的信息,再现三维物体形状的反问题.基于将在有限个测量方向上的远场数据作加权合成的思想,提出了求解这类反问题的一种简单快速算法.在这个算法中,既不需预先知道散射体的边界类型,也不需预先知道散射体个数的信息.对有限孔径远场数据的数值算例表明了这一方法是实用和有效的.     

Equivalent source method for identification of panel acoustic contribution in irregular enclosed sound field

For the interior sound field formed by the complex vibrating structure,an identification approach of panel acoustic contribution based on equivalent source method(ESM)was presented.The normal velocity on the surface of vibrating structure was first reconstructed by using interior nearfield acoustic holography based on ESM and the prediction of whole interior enclosed sound field was realized.Then the sound pressure produced by each panel at the interested field point was respectively replaced by the radiated pressure of the enclosed interior sound field which is formed by the equivalent virtual sources located near the surface of the cavity.Combining with the reconstructed normal surface velocity,the acoustic contribution of each panel to any position in the cavity was obtained by transforming the complex enclosed non-free field into the simple interior free field.Numerical simulations and experiments are conducted,and the influences of the number of the equivalent sources and the distance between them and the reconstructed surface have been investigated.The results show that the proposed method is easier to be implemented with the same accuracy than the traditional analysis method.     

复杂封闭声场面板声学贡献度识别的等效源法

针对由复杂结构振动形成的封闭空间声场,提出了一种基于等效源法的面板声学贡献度分析方法。该方法首先利用基于等效源法的内部声全息技术,重构出振动结构表面的法向振速并实现对整个内部封闭声场的预测。再将振动结构的每个面板在腔体内部场点产生的声压分别用位于空腔表面附近的等效源在该点产生的辐射声压代替,将复杂的封闭非自由声场问题转化为简单的内部自由场问题,结合重建出的结构表面法向振速进而识别出封闭振动结构各面板对腔体内任意位置的声学贡献度。通过对复杂结构内声场的数值仿真和验证实验,分析了等效源的数量及与重建面距离等参数对重建精度的影响,结果表明所提方法不仅能够达到传统数值分析方法的计算精度,而且具有更简单的求解过程。     

基于声源预估的波叠加组合全息声源识别方法

为解决声源识别中旁瓣效应等因素引起的虚假识别结果问题,提出了基于声源位置预估的波叠加组合全息算法。应用波束形成算法进行声源位置的预估,将预估声压峰值点作为假定声源点,通过波叠加原理建立方程组,利用声源强度求解结果构造完整全息平面,通过声全息方法重新计算重建面声压。进行了仿真和实验验证,经该算法重构后,波束形成算法产生的虚假声源处的声压得以明显降低。表明该算法可以抑制旁瓣效应等因素的影响,提高声源识别的准确率。