The radiating part of circular sources

An analysis is developed linking the form of the sound field from a circular source to the radial structure of the source, without recourse to far-field or other approximations. It is found that the information radiated into the field is limited, with the limit fixed by the wavenumber of the source multiplied by the source radius (Helmholtz number). The acoustic field is found in terms of the elementary fields generated by a set of line sources whose form is given by Chebyshev polynomials of the second kind and whose amplitude is found to be given by weighted integrals of the radial source term. The analysis is developed for tonal sources, such as rotors, and for Helmholtz number less than two, for random disk sources. In this case, the analysis yields the cross-spectrum between two points in the acoustic field. The analysis is applied to the problems of tonal radiation, random source radiation as a model problem for jet noise, and to noise cancellation, as in active control of noise from rotors. It is found that the approach gives an accurate model for the radiation problem and explicitly identifies those parts of a source which radiate.     

The time domain response of some systems for sound reproduction

The perception in a listener of the existence of a “virtual” source of sound at a prescribed spatial position can be produced by ensuring that the acoustic signals at the listener's ears faithfully replicate those that would be produced by a “real” source at the same position. When loudspeakers are used to transmit the signals, it is necessary to pass the signals intended for presentation at the listener's ears through a matrix of filters that provide the inverse of the matrix of transfer functions that relates the loudspeaker input signals to the listener's ear signals. The characteristics of such filter matrices are profoundly influenced by the conditioning of the matrix to be inverted. This filter design problem is reviewed here by representing the loudspeakers as simple point monopole sources the head of the listener as a rigid sphere. The case of a virtual acoustic imaging system that uses two loudspeakers in order to reproduce the signals at the two ears is first described in some detail and previous work is reviewed. It is confirmed that the time domain response of the reproduced field is of long duration at frequencies where the inversion problem is ill-conditioned. The influence of the presence of the listener's head on this time domain behaviour is also evaluated. The principle is then extended to four input-four output reproduction systems and the computational model is used to explain some previous experimental observations. Finally, the conditioning of five input-four output systems is also examined and shown to have some potentially desirable characteristics.     

Optimal design of loudspeaker arrays for robust cross-talk cancellation using the Taguchi method and the genetic algorithm

An optimal design technique of loudspeaker arrays for cross-talk cancellation with application in three-dimensional audio is presented. An array focusing scheme is presented on the basis of the inverse propagation that relates the transducers to a set of chosen control points. Tikhonov regularization is employed in designing the inverse cancellation filters. An extensive analysis is conducted to explore the cancellation performance and robustness issues. To best compromise the performance and robustness of the cross-talk cancellation system, optimal configurations are obtained with the aid of the Taguchi method and the genetic algorithm (GA). The proposed systems are further justified by physical as well as subjective experiments. The results reveal that large number of loudspeakers, closely spaced configuration, and optimal control point design all contribute to the robustness of cross-talk cancellation systems (CCS) against head misalignment.     

Spatial resolution limits for the reconstruction of acoustic source strength by inverse methods

One method for deducing the strength of an acoustic source distribution from measurement of the radiated field involves the inversion of the matrix of frequency response functions relating the field measurement points to the strengths of a number of point sources used to represent the source distribution. In practice, the frequency response function matrix to be inverted may very often be ill-conditioned. This ill-conditioning will also often result in an ill-posed problem and thus regularization algorithms are used to produce reasonable solutions. For this purpose, Tikhonov regularization has been applied, and generalized cross-validation (GCV) has been introduced as an effective method for determining the proper amount of regularization without prior knowledge of either the source distribution or the contaminating errors. In the present work, the emphasis is placed on the relationship between the spatial resolution of the reconstructed source distribution and the small singular values of the frequency response function matrix to be inverted. However, the use of Tikhonov regularization often suppresses the effect of small singular values and these are in turn often associated with high spatial frequencies of the source distribution. Thus, the process of regularization produces a useful estimate of the acoustic source strength distribution but with a limited spatial resolution. Furthermore, in the field of Fourier acoustics, the spatial resolution of the reconstructed source distribution is usually limited by the wavelength of the radiation. This paper expresses the relationship between estimation accuracy, spatial resolution, noise-level and source/sensor geometry, when a range of inverse sound radiation problems are regularised using Tikhonov regularization with GCV. The results presented form the basis of guidelines that enable the reconstruction of acoustic source strength with a resolution that is finer than the intrinsic half-wavelength limit.     

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。     

Stereophonic channel decorrelation using a binaural masking model

Traditional methods often only use monaural masking models to decorrelate input signals for stereo acoustic echo cancellation. Whereas, it seems more reasonable to use binaural masking models for the following two reasons. First, stereo signals are heard by two ears rather than just one. Second, psychoacoustic researchers have already shown that there are obvious masking level differences between binaural masking models and monaural masking models. By studying binaural masking level difference models, we first introduce a simplified binaural masking model for stereo acoustic echo cancellation. Considering that the interaural time difference is dominant at low frequencies (?$?$1.5  kHz) and the interaural level difference is a major cue at higher frequencies, we propose to use different signal decorrelation schemes at these two frequency bands. In the low-frequency band, a pitch-driven sinusoidal injection scheme is proposed to maintain the interaural time difference, where the amount of injection is determined by the proposed binaural masking model. In the high-frequency band, a modified sinusoidal phase modulation scheme is applied to make a trade-off between preserving the interaural level difference and decorrelating the stereophonic input signals. Assessment results show that the proposed method can effectively improve the non-unique problem and retain good speech quality.     

Effect of sensor proximity over the non-conformal hologram plane in the near-field acoustical holography based on the inverse boundary element method

Near-field acoustical holography (NAH) is a useful tool for the identification and visualization of vibro-acoustic sources. In particular, NAH can be applied to many practical sources having irregular shape if the inverse boundary element method (BEM) is employed. Once the relation between the source and the radiated field is defined in the transfer matrix modeled by the BEM, the reconstruction of acoustic parameters on the source surface can be conducted by the multiplication of the inverse transfer matrix and the field data measured over the hologram surface. The usual practical way to measure the field data radiated from an irregular shaped source is to adopt a regular hologram surface, which can be a flat, cylindrical, or spherical shape, for the measurement ease. Then, the hologram surface is not conformal to the source surface and the resulting transfer matrix becomes further ill-posed than the conformal case. To investigate the effect of sensor proximity and distance variability on the reconstruction error, simulation and measurement were conducted for an interior problem comprised of a parallelepiped rigid box with a vibrating end plate. Flat, tilted, and randomized hologram surfaces were adopted in the test. It was shown that the reconstruction error is greatly affected by the conditioning of transfer matrix which is related to the positioning of sensor. To improve the reconstruction accuracy, an investigation was conducted to find proper field points among a large number of overdetermined field points. The number of field data was reduced gradually under various reduction schemes using condition number, effective independence value, and sensor distance. It was demonstrated that the quality of reconstruction result given by the non-conformal measurement can be improved by removing some field points that contribute to the ill-conditioning of the inverse problem. A small improvement of the reconstruction accuracy was observed by reducing the field points in the overdetermined situation. However, further reduction of the field points, becoming an underdetermined situation, yielded a drastic improvement of the reconstruction accuracy.     

Analyzing and correcting spectrometer temperature sensitivity

Stability and reproducibility of the spectrometer are fundamental to the success of many modern NMR experiments. Variation in room temperature is a particularly important source of instability, in part because it can cause coherent artifacts in NMR spectra. Small changes in room temperature lead to corresponding changes in the phases, amplitudes, and frequencies of NMR signals. These can lead in turn to apparently random spectral artifacts such as t(1)-noise in two-dimensional (2D) NMR and to the incomplete cancellation of signals in difference spectra, but also, importantly, to F(1) satellite signals in 2D spectra. These "parallel diagonals" arise from the use of air conditioning, which typically forces room temperature to oscillate within a fixed band. Work to identify, quantify, and suppress sources of temperature sensitivity in a modern 300-MHz spectrometer has led to a greater than 10-fold improvement in the signal-to-artifact ratio.     

Physical modeling of active cancellation of low-frequency sound signals

Some new methods of active cancellation of low-frequency sound signals have been developed in view of the low efficiency of passive methods at these frequencies. Examples of developing algorithms and technical tools for active suppression of sound signals in air and water are presented. The solution of this problem is of paramount importance for reducing noise and providing ecological safety in transport, aviation, and shipbuilding industry. A real possibility of suppressing the low-frequency discrete components by no less than 10 to 15 dB and reducing the sound level in a wide frequency band by no less than 6 to 8 dB is demonstrated. A wide range of applications for the technology of active cancellation of discrete components and noise is indicated.     

Spatial correlation of sound fields from underwater explosions in the Barents Sea

Results of an experimental study of the spatial correlation of signals from underwater explosions are presented for vertically and horizontally (across the path) separated reception points. The measurements were carried out on two propagation paths about 200 and 110 km long in the Barents Sea for frequencies below 500 Hz. The vertical correlation radius is found to increase with the distance to the source, which is primarily caused by the decrease in the sector of arrival angles of energy carrying signals. The transverse correlation radius weakly depends on the path length for frequencies below 300 Hz, while for higher frequencies, it decreases beginning from a distance of about 40 km, which can be attributed to the effect of short-period internal waves.     

The influence of signal parameters on the sound source localization ability of a harbor porpoise (Phocoena phocoena)

It is unclear how well harbor porpoises can locate sound sources, and thus can locate acoustic alarms on gillnets. Therefore the ability of a porpoise to determine the location of a sound source was determined. The animal was trained to indicate the active one of 16 transducers in a 16-m-diam circle around a central listening station. The duration and received level of the narrowband frequency-modulated signals (center frequencies 16, 64 and 100 kHz) were varied. The animal's localization performance increased when the signal duration increased from 600 to 1000 ms. The lower the received sound pressure level (SPL) of the signal, the harder the animal found it to localize the sound source. When pulse duration was long enough (approximately 1 s) and the received SPLs of the sounds were high (34-50 dB above basic hearing thresholds or 3-15 dB above the theoretical masked detection threshold in the ambient noise condition of the present study), the animal could locate sounds of the three frequencies almost equally well. The porpoise was able to locate sound sources up to 124 degrees to its left or right more easily than sounds from behind it.     

单层传声器阵列重建相干声场的近场声全息方法研究

针对传声器阵列两侧存在相干声源的非自由声场重建问题,提出基于球面谐波函数扩展近场声全息理论的相干声场重建方法。该方法在已知测量面两侧声源几何位置时,使用单层传声器阵列获取测量面处的声压分布,通过最小二乘法获得与目标声源和干扰噪声源响应对应的最优球波函数扩展项数和最优系数向量,结合测点位置的空间坐标进行声波分解,并分别重建出各声源在测量面上的声压分布。为了验证方法的有效性,分别给出了相干噪声源为球形声源和非球形声源的仿真验证,并在全消声室内对双扬声器产生的相干声场的重建进行了实验验证。结果表明:该方法对球形声源和非球形声源干扰下的声场重建都具有较好的效果,球形声源干扰下的重建精度更高。      

Regularization for improving the deconvolution in real-time near-field acoustic holography

Near-field acoustic holography is a measuring process for locating and characterizing stationary sound sources from measurements made by a microphone array in the near-field of the acoustic source plane. A technique called real-time near-field acoustic holography (RT-NAH) has been introduced to extend this method in the case of nonstationary sources. This technique is based on a formulation which describes the propagation of time-dependent sound pressure signals on a forward plane using a convolution product with an impulse response in the time-wavenumber domain. Thus the backward propagation of the pressure field is obtained by deconvolution. Taking the evanescent waves into account in RT-NAH improves the spatial resolution of the solution but makes the deconvolution problem "ill-posed" and often yields inappropriate solutions. The purpose of this paper is to focus on solving this deconvolution problem. Two deconvolution methods are compared: one uses a singular value decomposition and a standard Tikhonov regularization and the other one is based on optimum Wiener filtering. A simulation involving monopoles driven by nonstationary signals demonstrates, by means of objective indicators, the accuracy of the time-dependent reconstructed sound field. The results highlight the advantage of using regularization and particularly in the presence of measurement noise.     

Simple non-invasive measurement of rapid eye vibration

A simple, non-invasive method for the measurement of eye vibrations above 30 Hz is described. The method can be used in either laboratory or natural conditions, and is based on the cancellation of an illusion of motion that occurs when two nearby light sources flickering in counterphase above the flicker fusion limit are observed during eye vibration. In these conditions, the light sources appear to oscillate in space at a frequency equal to the difference between the vibration and flicker frequencies. The frequency of eye vibration can be determined by adjusting the flicker frequency until the illusion disappears (i.e., until the difference frequency becomes zero). The same set-up can also be used to determine the amplitude of eye vibration, by adjusting the spatial separation between the two light sources until the oscillation appears to be the result of their bouncing off each other upon contact. The reliability and sensitivity of this method are illustrated with data from three observers whose eyes were vibrated with a commercial massager applied onto their neck, and using three different settings for the speed of the massager.     

On the dependence of (f2-f1) difference tones on subject and on additional masker

For a group of eight subjects showing stronger than usual irregularities in the level dependence of the quadratic distortion product, the level and phase of the (f2-f1) difference tone were measured using the method of cancellation for three sets of primary frequencies as a function of the primary levels. An additional masker seems to "linearize" the level dependence toward regular behavior. Using data sets produced with and without the additional masker, it is possible to separate two sources of quadratic nonlinearity, one with regular behavior presumably located in the middle ear and another with irregular behavior (similar to that of cubic distortion) presumably located in the characteristics of outer hair cells. Through the "subtraction" of empirically determined patterns from idealized patterns, it is possible to approximate patterns stemming from the inner-ear source alone.     

封闭空间声场重构的多层等效源法

对于封闭空间内的多途反射声,传统的等效声源法将其等效为距离边界一定距离的单层等效声源体进行声场重构,然而等效源与边界的距离选取依据不确定。因此,为获得等效声源配置的最优距离,在等效声源法(ESM)的基础上构建多层等效声源,提出一种适用于封闭环境声场重构的多层等效声源法(MESM),并依据等效声源的空间分布的稀疏性来获得等效声源强度信息。首先给出多层等效源法的理论依据,其次通过数值计算以及实验测试两种方式对比验证了所提方法。数值结果表明:MESM相比于ESM可在600 Hz以上频段获得低5~10 dB左右的重构误差,但是200 Hz以下的低频重构误差会增加5 dB左右。实验结果表明:MESM可比ESM获得更低的重构误差。文章最后基于数值计算研究了所提方法的主要影响因素。研究表明:虽然MESM会比ESM耗费2倍的计算时间,但在整体频率范围内,MESM可在ESM基础上提升600 Hz以上的重构性能。另外,等效声源的层数和层内数目的改变不会影响声场重构性能,而当传声器数目较多、阵列位置随机、空间边界的吸声系数不是很大时,MESM可获得比ESM更低重构误差,特别是600 Hz以上的中频段区间。      

Front-back confusion resolution in three-dimensional sound localization using databases built with a dummy head

Sound localization plays an important role in everyday life. It helps us to separate sounds coming from different sources and thus to acquire acoustic information. This paper describes an algorithm for localizing the position of a sound source, as recorded by dummy head microphones. The recorded signals are considered to be basic, random signals within an imaginary round room. The goal of this research is to localize random signals produced from different positions using information about basic signals. The method used is based on the identification of similarities between basic and random signals. It includes an interaural time difference comparison at the beginning, and continues with further analysis of the differences in signal spectrums. One of the main issues arising in sound localization is the problem of front-back confusion, and this paper shows how it was resolved by the use of reference signals.     

Sound cancellation by the use of secondary multipoles: experiments

Theory related to global, free-field cancellation of a primary monopole field by the use of a displaced, secondary multipole was presented previously: a corresponding experimental investigation is presented here. The construction of multipole source components to octopole order is described, as are procedures for determining their source strengths. Dipoles, longitudinal quadrupoles, and longitudinal octopoles that conformed closely to their theoretical models were constructed using arrays of unbaffled loudspeakers. Two methods of calculating the multipole strengths required to cancel a primary monopole field were implemented in an open-loop manner: a "direct" approach based on a multipole expansion of the primary field, and a least-squares procedure based on fitting the secondary field to the primary field either along a circle enclosing a secondary source, or along a segment of that circle. Cancellation measurements were made on a 1-m-radius circle centered on the secondary source: the primary-to-secondary source separation was approximately 0.2 wavelengths. It was found both that a secondary multipole could provide far greater cancellation than a monopole placed at the same distance from the primary source and that the least-squares approach resulted in greater far-field cancellation than did the direct approach.     

A Novel Circuit Architecture for High Performance of High-order Subharmonic Mixers

A novel circuit architecture for high performance of high-order subharmonic (SH) mixers is proposed in this paper. According to the specified harmonic mixing order, one or more mixer diodes sub-arrays and corresponding power divider as well as phase shift network for RF and LO signals are arranged in the circuit. This proposed SH mixer circuit has improved conversion loss, wide dynamic range and high port isolation for high-order SH mixers. By phase cancellation of idle frequencies, the proposed SH mixer circuit can eliminate complicated design procedure of idle frequency circuits; by phase cancellation of leakage LO power to RF and IF port, and leakage RF power to LO port, the mixer circuit can get high port isolation in LO-IF/RF and RF-LO. The increased antiparallel diode pairs in each sub-array will also lead to well performance by lowering effective series resistance. The proposed SH mixer circuit can be easily realized with power divider and phase shift network for RF and LO signals. Supported in part by National Natural Science Foundation of China (NSFC) under Grants 60621002 and in part by NSFC under Grants 60471017.     

Time-delay estimation of acoustic emission signals using ICA

Acoustic emission (AE) analysis is used for characterization and location of developing defects in materials. AE sources often generate a mixture of various statistically independent signals. One difficult problem of AE analysis is the separation and characterization of signal components when the signals from various sources and the way in which the signals were mixed are unknown. Recently, blind source separation by independent component analysis (ICA) has been used to solve these problems. The main purpose of this paper is to demonstrate the applicability of ICA to time-delay (T-D) estimation of two independent continuous AE sources on an aluminum beam. It is shown that it is possible to estimate T-Ds by ICA, and thus to locate two independent simultaneously emitted sources.