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。     

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.     

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 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。     

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.     

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。     

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.     

Multichannel time-reversal processing for acoustic communications in a highly reverberant environment

The development of time-reversal (T/R) communication systems is a recent signal processing research area dominated by applying T/R techniques to communicate in hostile environments. The fundamental concept is based on time-reversing the impulse response or Green's function characterizing the uncertain communications channel to mitigate deleterious dispersion and multipath effects. In this paper, we extend point-to-point to array-to-point communications by first establishing the basic theory to define and solve the underlying multichannel communications problem and then developing various realizations of the resulting T/R receivers. We show that not only do these receivers perform well in a hostile environment, but they also can be implemented with a "1 bit" analog-to-digital converter design structure. We validate these results by performing proof-of-principle acoustic communications experiments in air. It is shown that the resulting T/R receivers are capable of extracting the transmitted coded sequence from noisy microphone array measurements with zero-bit error.     

A new method for calculating potential curves in the zero-order approximation of a generalized WKB method

A new method, different from that of [1 ], is proposed for plotting potential curves from vibrational-rotational levels in the zero-order approximation in Planck's constant of a generalized WKB method.Translated from Izvestiya Vysshikh Uchebnykh Zavedenii, Fizika, Vol. 16, No. 7, pp. 127–129, July, 1973.     

Optimal design method for magnetization directions of a permanent magnet array

In many magnetic systems, the permanent magnet (PM) pattern has a great influence on their performance. This study proposes a systematic optimization method for designing discrete magnetization directions. While previous works have been mostly dependent on researchers’ intuition, the developed method is systematic and can be applied to a two-dimensional PM-type eddy current brake model. The effectiveness of the method is confirmed, where the design’s aim is to maximize the braking force on a moving conductor. The sensitivity analysis is accomplished by the adjoint variable method and the sequential linear programming is used as an optimizer. Several optimization results for various conditions through the proposed design method are compared to each other and the optimal magnet configuration for an eddy current brake is suggested.     

A nonlinear dynamics method for signal identification

When a radio frequency signal is radiated by a transmitter, the properties of the transmitter itself affect the properties of the signal. These transmitter-induced changes are known as unintentional modulation, to differentiate them from intentional modulation used to add information to the signal. The unintentional modulation can be used to identify which transmitter produced a signal. This paper shows how phase space analysis based on nonlinear dynamics ideas can be used to determine which of several amplifiers produced a signal.     

A new method of charged particle identification based on frequency spectrum analysis

A new frequency domain method for charged particle identification, called Frequency Ratio Analysis(FRA), is proposed by analyzing the frequency spectra of proton pulses and alpha pulses acquired from a totally depleted Si detector. Identification performance of the FRA method is evaluated and compared with two time domain methods, the current pulse amplitude method and the second moment method. The results show that the FRA method is not only feasible and effective but also superior to the two time domain methods, as it achieves an obvious increase in value of the figure-of-merit(FOM).     

Time-reversal processing for an acoustic communications experiment in a highly reverberant environment

Time-reversal (T/R) communications is a new application area motivated by the recent advances in T/R theory. Although perceived by many in signal processing as simply an application of matched-filter theory, a T/R receiver offers an interesting solution to the communications problem for a reverberant channel. In this paper, the performance of various realizations of the T/R receiver for an acoustic communications experiment in air is described along with its associated processing. The experiment is developed to evaluate the performance of point-to-point T/R receivers designed to extract a transmitted information sequence propagating in a highly reverberant environment. It is demonstrated that T/R receivers are capable of extracting the transmitted coded sequence from noisy microphone sensor measurements with zero-symbol error. The processing required to validate these experimental results is discussed. These results are also compared with those produced by an equivalent linear equalizer or inverse filter, which provides the optimal solution when it incorporates all of the reverberations.