Adaptive wave field synthesis for active sound field reproduction: experimental results

Sound field reproduction has applications in music reproduction, spatial audio, sound environment reproduction, and experimental acoustics. Sound field reproduction can be used to artificially reproduce the spatial character of natural hearing. The objective is then to reproduce a sound field in a real reproduction environment. Wave field synthesis (WFS) is a known open-loop technology which assumes that the reproduction environment is anechoic. The room response thus reduces the quality of the physical sound field reproduction by WFS. In recent research papers, adaptive wave field synthesis (AWFS) was defined as a potential solution to compensate for these quality reductions from which WFS objective performance suffers. In this paper, AWFS is experimentally investigated as an active sound field reproduction system with a limited number of reproduction error sensors to compensate for the response of the listening environment. Two digital signal processing algorithms for AWFS are used for comparison purposes, one of which is based on independent radiation mode control. AWFS performed propagating sound field reproduction better than WFS in three tested reproduction spaces (hemianechoic chamber, standard laboratory space, and reverberation chamber).     

A wave field synthesis approach to reproduction of spatially correlated sound fields

This article discusses an open-loop wave field synthesis (WFS) approach for the reproduction of spatially correlated sound fields. The main application concerns laboratory reproduction of turbulent boundary layer wall pressure on aircraft fuselages and measurement of their sound transmission loss. The problem configuration involves reconstruction of random sound pressure distributions on a planar reproduction surface using a planar array of reproduction monopoles parallel to the reproduction plane. In this paper, the WFS formulation is extended to sound fields with imposed time and spatial correlation properties (or equivalently imposed cross-spectral density in the frequency and wave number domains). Numerical examples are presented for the reproduction of a propagating plane wave, diffuse acoustic field and wall pressure in subsonic or supersonic turbulent boundary layers. The reproduction accuracy is examined in terms of the size of the source plane and reproduction plane, their separation, and the number of reproduction sources required per acoustic wavelength. While the reproduction approach cannot reconstruct sub-wavelength correlation scales of subsonic turbulent boundary layers, it effectively reconstructs correlation scales larger than the acoustic wavelength, making it appropriate for diffuse acoustic field and supersonic turbulent layers.     

Wave field synthesis of a sound field described by spherical harmonics expansion coefficients

Near-field compensated higher order Ambisonics (NFC-HOA) and wave field synthesis (WFS) constitute the two best-known analytic sound field synthesis methods. While WFS is typically used for the synthesis of virtual sound scenes, NFC-HOA is typically employed in order to synthesize sound fields that have been captured with appropriate microphone arrays. Such recorded sound fields are essentially represented by the coefficients of the underlying surface spherical harmonics expansion. A sound field described by such coefficients cannot be straightforwardly synthesized in WFS. This is a consequence of the fact that, unlike in NFC-HOA, it is critical in WFS to carefully select those loudspeakers that contribute to the synthesis of a given sound source in a sound field under consideration. In order to enable such a secondary source selection, it is proposed to employ the well-known concept of decomposing the sound field under consideration into a continuum of plane waves, for which the secondary source selection is straightforward. The plane wave representation is projected onto the horizontal plane and a closed form expression of the secondary source driving signals for horizontal WFS systems of arbitrary convex shape is derived.     

Two-photon Jaynes-Cummings systems interacting with a classical field

If a two-level atom is in the two-photon resonance with a quantized mode and simultaneously inter-acts with a quasi-resonance classical field, then a photon exchange is observed in this system between the quantized and classical modes. It is demonstrated that such a physical system can serve as a source of squeezed radiation in the quantized mode. The squeezing can be arbitrarily close to unity, while the radiation amplitude can be relatively large. A situation is discussed when N atoms are in the two-photon resonance with a quantized mode and simultaneously interact with a classical field. The phenomenon of exponential superradiation is described when the number of photons in the quantized mode exponentially depends on the number N of atoms.     

Modified physical optics algorithm for near field scattering

A novel modified physical optics algorithm is proposed to overcome the difficulties of near field scattering prediction for classical physical optics. The method is applied to calculating the near field radar cross section of electrically large objects by taking into account the influence of the distinct wave propagation vector, the near field Green function, and the antenna radiation pattern. By setting up local reference coordinates, each partitioned facet has its own distinct wave front curvature. The radiation gain for every surface element is taken into consideration based on the modulation of the antenna radiation pattern. The Green function is refined both in amplitude and phase terms and allows for near field calculation. The scattered characteristics of the near field targets are studied by numerical simulations. The results show that the approach can achieve a satisfactory accuracy.     

Adaptive processing of signals in antenna arrays using the maximum likelihood method

Using the maximum likelihood method, we derive and analyze the procedures of adaptive processing of signals in antenna arrays when the noise covariance matrix is unknown for case A where the time dependence of a useful signal is known with an accuracy of up to several numerical parameters and case B of completely unknown time dependence of a useful signal. For case A, the problem is shown to have a unique solution which is somewhat different from the standard procedure of processing of signals in adaptive antenna arrays. In case B, the solution is not unique. Additional conditions for choosing the unique solution are formulated. A comparison with the classical method of adaptive processing of signals in antenna arrays is proposed. Translated from Izvestiya Vysshikh Uchebnykh Zavedenii, Radiofizika, Vol. 52, No. 4, pp. 343–354, April 2009.     

Controlled sound field with a dual layer loudspeaker array

Controlled sound interference has been extensively investigated using a prototype dual layer loudspeaker array comprised of 16 loudspeakers. Results are presented for measures of array performance such as input signal power, directivity of sound radiation and accuracy of sound reproduction resulting from the application of conventional control methods such as minimization of error in mean squared pressure, maximization of energy difference and minimization of weighted pressure error and energy. Procedures for selecting the tuning parameters have also been introduced. With these conventional concepts aimed at the production of acoustically bright and dark zones, all the control methods used require a trade-off between radiation directivity and reproduction accuracy in the bright zone. An alternative solution is proposed which can achieve better performance based on the measures presented simultaneously by inserting a low priority zone named as the “gray” zone. This involves the weighted minimization of mean-squared errors in both bright and dark zones together with the gray zone in which the minimization error is given less importance. This results in the production of directional bright zone in which the accuracy of sound reproduction is maintained with less required input power. The results of simulations and experiments are shown to be in excellent agreement.     

Application of a modified gradient least squares algorithm to an adaptive,actively quenched,sound field system

A modified least squares algorithm, preventing the overflow of the discharge grid of weight coefficients of an adaptive transverse filter and guaranteeing stable system operation, is suggested for the tuning of an adaptive system of an actively quenched sound field. Experimental results are provided for an adaptive filter with a modified algorithm in a system of several harmonic components of an actively quenched sound field.Nizhegorod State University. Translated from Izvestiya Vyssikh Uchebnykh Zavedenii, Radiofizika, Vol. 37, No. 10, pp. 1262–1272, October 1994.     

重放自由场虚拟源距离信息的动态双耳Ambisonics方法

双耳重放的目标之一是在耳机重放中产生不同方向和距离的虚拟源感知。本文研究了动态双耳Ambisonics重放自由场虚拟源方向和距离信息的简化信号处理方法。该信号处理方法包括两步:第1步是基于目标声场的球谐函数分解,合成采用扬声器的近场Ambisonics重放中逐级重构目标声场的信号;第2步是采用虚拟扬声器重放的方法,用动态头相关函数滤波处理将Ambisonics的扬声器重放信号转换为双耳重放信号并用耳机重放。进一步研究了动态双耳Ambisonics的阶数对定位效果的影响,为简化信号处理提供依据。对重放产生的双耳声压分析表明,5阶动态双耳Ambisonics重放足以提供听觉方向定位和距离感知的重要信息。同时心理声学的实验结果表明,结合声源距离相关的响度因素,5阶动态双耳Ambisonics重放可产生不同方向和1.0 m以下不同近场距离的自由场虚拟源的听觉感知。本文的方法仅需要固定距离的48个均匀空间方向的远场非个性化HRTF处理,实现了信号处理的简化。      

Improvement of Shack-Hartmann wave-front sensor measurement for extreme adaptive optics

The development of high-performance adaptive optics systems requires the optimization of wave-front sensors (WFSs) working in the high-order correction regime. We propose a new method to improve the wave-front slope estimation of a Shack-Hartmann WFS in such a regime. Based on a detailed analysis of the different errors in the slope estimation with a classical centroid and with the new method, the gain in terms of wave-front-sensing accuracy in both the detector and the photon noise regimes is stressed. This improvement is proposed without major system disruption.     

波场合成与波场分析的有源房间补偿方法

波场合成是一种空间声重放技术,利用扬声器阵列在宽阔的听音区域内重建声场。为消除或者消减听音房间的反射对重建波场的影响,利用测量环绕听音区域的闭合曲线上声压和声压梯度来分析波场,推导了基于圆形阵列进行波域分解的公式,利用多通道逆滤波进行了平面波域的房间补偿,实验结果显示该算法在整个听音区域内都是有效的。与传统补偿方法相比,边界元法所需的测量传声器数目少计算复杂度低,而波域分解具有更充分的波场分析能力,因此是一种更有效的有源房间补偿方法.      

基于 L-曲线参数优化的均匀声场重建算法

为了得到最小二乘法声场重建问题的稳定解,通常需要引入Tikhonov正则化方法。然而正则化程度取决于正则化参数的选择。针对这一问题,提出了一种基于L-曲线法参数选择的均匀声场重建算法。该算法根据重建误差与扬声器功率计算得到L-曲线,该曲线上曲率最大的点所对应的参数值作为Tikhonov正则化参数的选值。确定正则化参数后可进一步得到扬声器权系数以及重建均匀声场。针对不同正则化参数取值方法,对控制区域进行均匀声场重建以及重建性能仿真。仿真结果及实验表明,L-曲线法实现了重建误差与扬声器驱动信号功率之间的平衡。     

近场子空间聚焦的碰摩故障声发射定位方法

针对宽带多源声发射信号的相干、多模态和能量衰减快问题,提出一种近场多重相干信号子空间聚焦的定位算法用于碰摩故障声发射源的定位检测。首先,为滤除干扰模态波、减小频散效应,采用基于模态声发射传播特性分析的小波分解滤波方法,从碰摩初期的声发射信号中获取零阶模态波及波速用于定位计算;其次,为实现信号解相干,提出基于双边相关变换(TCT)的近场聚焦矩阵估计方法;最后,针对声发射信号的能量衰减快问题,利用近场基于特征分解的多重信号分类(N-MUSIC)的空间谱估计方法来实现声源的精确定位。理论分析和实验结果表明:该方法定位精度高、计算复杂度低、稳定性强,能有效识别多个相干碰摩声源。相比传统相干子空间算法(CSM),该方法减少了信号初值和聚焦频点的计算量,对双声源的分辨概率较现有修正近场多重信号分类算法提高了17%,是一种有效的碰摩故障源检测方法。      

Analysis of the amplitude and phase structure of transmitting media with probing field registration in the image plane

The problem of obtaining information on the amplitude and phase internal structure of a medium in which radiation propagates is considered. The information is extracted by probing the medium; the information on the amplitude and phase distribution of the probing field behind the transmitting medium in the plane of image formation is analyzed. A modified version of the modulation-spectral method proposed earlier by the authors is applied. In this version, there is no need to act on the probing field in the plane under investigation. The interpretation of results is simplified since the image is registered. Two versions of the schematic solution are analyzed. The first version corresponds to the experimental scheme intended for media that produce a modulating action on radiation and is described by multiplication by a complex function characterizing the action. The second version corresponds to the case when the action of the medium leads to a redistribution of radiation and can be presented by the convolution of the probing signal and the function describing the action.     

横向电场对平面织构胆甾相液晶螺距及反射特性的影响

阐述一种理论模型,讨论共面转换（IPS）模式下横向电场对平面织构胆甾相液晶螺距及扭曲角的影响．利用琼斯矩阵法计算了光正入射情况下未加电场时胆甾相液晶的反射谱．在上述模型基础上,计算了施加电场后胆甾相液晶的反射峰值波长以及反射带宽随场强的变化关系．探讨了忽略锚定与强锚定两种边界条件下横向电场对平面织构胆甾相液晶螺距及反射特性的影响．所得结论在理论上证实：共面转换模式下电场可以调谐胆甾相液晶的反射光颜色,从而为基于电控螺距原理的胆甾相液晶反射式彩色显示方案提供了理论上的依据． 关键词： 胆甾相液晶 电控螺距 共面转换 琼斯矩阵法     

海面偶极子声场中近程运动参数估计

对海面偶极子声场提出基于单个水听器的定位算法,并设计信号处理流程和双曲线提取的Hough变换法。海试结果表明信号处理和双曲线提取具有较高增益,定位算法稳健、具有较高精度。该算法所需设备简单,运算量小,在近距离目标运动参数估计和辐射噪声测量方面具有特色。     

外视场拼接测量系统的视场拼接和交汇测量算法及其实现

基于2台外视场拼接高速电视测量仪原型样机（硬件）提出了一种将交汇测量和拼接处理相结合的算法。根据需要对地球质心和光电测量系统建立了5个坐标系,介绍了坐标系的定义及其它们之间的变换过程,给出了目标轨迹交汇测量和视场拼接的实例。用2台高速电视测量仪拼接的8台测量相机同时对同一运动目标进行测量,对获得的测量数据进行交汇和视场拼接处理,结果显示提出的拼接算法是正确、有效的,可以得到唯一解;地球曲率半径和地球子午线收敛两项影响因素得到了完全修正,表明交汇测量算法完全可以推广到其它光电测控仪器的交汇测量。     

Wave field synthesis of moving virtual sound sources with complex radiation properties

An approach to the synthesis of moving virtual sound sources with complex radiation properties in wave field synthesis is presented. The approach exploits the fact that any stationary sound source of finite spatial extent radiates spherical waves at sufficient distance. The angular dependency of the radiation properties of the source under consideration is reflected by the amplitude and phase distribution on the spherical wave fronts. The sound field emitted by a uniformly moving monopole source is derived and the far-field radiation properties of the complex virtual source under consideration are incorporated in order to derive a closed-form expression for the loudspeaker driving signal. The results are illustrated via numerical simulations of the synthesis of the sound field of a sample moving complex virtual source.     

An innovation based random weighting estimation mechanism for denoising fiber optic gyro drift signal

In Interferometric Fiber Optic Gyroscope (IFOG), the diminution of random noise and drift error is a critical task. These errors degrade the performance of IFOG. In this paper, a modified adaptive Kalman gain correction (AKFG) algorithm is proposed to denoise IFOG signal. The covariance matrix of innovation sequence is estimated using weighted average window method in which the weights are randomly generated in the range [0, 1]. Innovation based random weighted estimation (IRWE)-AKFG is applied to denoise the IFOG drift signal. The Kalman gain is adaptively updated using the covariance matrix of innovation sequence. The proposed algorithm is applied for denoising IFOG signal under static and dynamic environment. Allan variance method is used to analyze and quantify the stochastic errors in IFOG sensor. The performance of the proposed algorithm is compared with Conventional Kalman filter (CKF) and the simulation results reveal that the proposed algorithm is an efficient algorithm for denoising the IFOG signal.