封闭空间声场的虚源法模拟研究

文中以一个不规则的空间为例,给出了虚源法模拟声场的基本原理和算法,介绍了由模拟所得的脉冲响应获取声场明晰度等参数的方法。针对虚源法计算量巨大这一突出问题,着重讨论了几种提高计算效率的方法,并在最后将虚源法与声线跟踪法作了简单比较。     

声音制作中的虚拟技术

本文介绍了用于广播、影视节目声音制作的虚拟技术的基本原理和技术关键，包括实在场景声场脉冲响应函数的测量方法，想象场影声场计算机模拟方法、虚拟处理的实施等问题。并对其中存在的问题和有待进一步研究和领域做了初步讨论。     

封闭声场数值模拟和预测

以声线跟踪法为基础,利用Visual C 和OpenGL研制了一个可用于声场模拟和预测的计算机仿真程序SOFIS（Sound Field Simulation）。它利用计算机对声场中的各要素,包括声源、声传播方式及接收者等加以模拟,由此求出声场中不同点处的脉冲响应,并能计算对应的若干声学参数（如明晰度）。文中详细给出了SOFIS预测房间音质的原理和方法,并通过实例对其可行性进行了论证。     

音乐厅声场的计算机模拟

１引言使用计算机进行室内声学模拟技术主要有两大类：一类是以波动方程为基础进行计算；一类是以几何声学为基础进行计算。前者主要包括有限元法和边界元法，这种方法对计算机性能要求高，且运行时间长，受到很大局限；后者主要包括声线跟踪法和虚声源法（声像法），该方法应用声线的反射关系，跟踪各条声线传播进程以及应用虚声源原理来确定房间各界面引起的声反射规律来研究、计算和模拟实际声场的特性。这种方法的计算过程较前者简单，因而得到了推广和应用。从１９５８年第一篇有关室内声场计算机模拟的论文公开发表以来，直到２０世纪…     

线形宽带超声相控阵换能器的声场分析

讨论了一种计算矩形活塞阵元声场脉冲响应函数的简便算法,通过对不同延时量的矩形活塞阵元声场脉冲响应函数进行叠加,得到线形换能器阵列的脉冲响应函数;再施以宽带脉冲激励函数,得到线形换能器阵列的声场分布。比较了3种不同换能器的脉冲响应及声场分布,该法对工业无损检测领域超声成像用换能器参数优化具有重要意义。     

脉冲响应函数在Monte Carlo模拟中的应用研究

MonteCarlo方法是研究复杂面光源入射条件下光在生物组织分布规律的唯一有效手段,但直接的MonteCarlo模拟存在计算量大、收敛速度慢的问题。通过引入生物组织脉冲响应函数的概念,利用卷积方法计算光在穿过组织一定深度后的分布规律,编制了相应的程序并进行了大量计算机模拟,在此基础上,提出了一种对模拟结果进行评价的方法,利用该方法分析了光源离散量与脉冲响应函数模拟量对光场分布计算结果的影响,证明基于脉冲响应函数的MonteCarlo模拟可以极大地提高计算效率和收敛速度。     

房间声场动态特性与主观音质参数的初步探讨

扬声器系统在听音室中位置的选择，目前国际上尚未有严格和明确的规定。如：IEC－268－13及IEC－543等只笼统地指出了一个区域，近年来，一些国外学者提出了这个问题，但仅限于在频域上进行了研究，我们通过实测室内声场，并用镜像法在计算机上进行模拟，得知房间中声场的频率响应随时间有很大变化，另一方面，通过主观试听，发现反射声的分布对人的听感有较大的影响，因此，时间因素的影响也是不可忽略的。为了判定反射声是否能造成可以听到的染色效应，我们求得房间脉冲响应的自相关函数，并通过计算“计权自相关函数”来判断染色产应能否被察觉，同时，为了反映室内声场的扩散性，并与双耳听觉联系起来，我们求出了双耳互相关孙数，由此来优选扬声器在听音室中的位置。     

室内可见光LED信道特性研究

为了研究室内多个白光LED构成的室内无线信道的特性,首先分析了多LED通信系统的基本构成,然后分析了光线追迹法求脉冲响应的基本原理和基本方法,提出了用简化的光线追迹法计算信道的脉冲响应,这种方法仅仅考虑三次以下的反射情况.最后,用数值仿真方法对室内信道的脉冲响应进行了研究.仿真结果表明,系统的脉冲响应以直接响应和一阶响应为主,并随接收器和发射装置的位置不同而发生变化.     

脉冲响应函数估计方法的改进

脉冲响应函数(IRF)的估计是一项有重大理论和实践意义的工作,但是,该项研究并未引起足够的重视,这与其拥有的价值很不相称。本文在分析频响函数(FRF)的实部和虚部特性基础上,提出用虚部变换取代经典的FRF付氏逆变换.数据模拟算例和试验数据处理表明,虚部变换法大幅度地提高了IRF的估计精度。而且,该方法还有效地提高了变换效率.     

EASE中％ALcons的计算方法

EASE中计算辅音清晰度损失百分率的Peutz公式有多种形式，基于扩散声场理论推导出不同形式的Peutz公式的计算方法，并计算了一个具体算例。理论推导和算例研究表明：EASE的标准模拟中的％ALcons计算值与理论推导值吻合：不同形式的Peutz公式之间存在一定关系。算例计算显示，EASE中AURA模块计算的％ALALcons值与理论推导值之间存在较大差异；相同形式Peutz公式在不同分割时间时计算值之间存在较大差异，同一分割时间不同形式Peutz公式的计算值之间也存在较大差异。     

Optimal Design of 2D FIR Filters with Quadrantally Symmetric Properties Using Fractional Derivative Constraints

In this article, an optimal design of two-dimensional finite impulse response (2D FIR) filter with quadrantally even symmetric impulse response using fractional derivative constraints (FDCs) is presented. Firstly, design problem of 2D FIR filter is formulated as an optimization problem. Then, FDCs are imposed over the integral absolute error for designing of the quadrantally even symmetric impulse response filter. The optimized FDCs are applied over the prescribed frequency points. Next, the optimized filter impulse response coefficients are computed using a hybrid optimization technique, called hybrid particle swarm optimization and gravitational search algorithm (HPSO-GSA). Further, FDC values are also optimized such that flat passband and stopband frequency response is achieved and the absolute \(L_1\)-error is minimized. Finally, four design examples of 2D low-pass, high-pass, band-pass and band-stop filters are demonstrated to justify the design accuracy in terms of passband error, stopband error, maximum passband ripple, minimum stopband attenuation and execution time. Simulation results have been compared with the other optimization algorithms, such as real-coded genetic algorithm, particle swarm optimization and gravitational search algorithm. It is observed that HPSO-GSA gives improved results for 2D FIR-FDC filter design problem. In comparison with other existing techniques of 2D FIR filter design, the proposed method shows improved design accuracy and flexibility with varying values of FDCs.     

面光源照射下生物组织中光传输规律的计算机模拟

本文利用基于脉冲响应函数的MonteCarlo方法,模拟了了不同分布的面光源照射条件下,生物组织中不同层面上的二维光场分布,分析了透射深度与入射光源分布对组织中光分布特点的影响,说明该模拟方法计算效率高且统计稳定性好,同时讲座了不同情况下,脉冲响应函数模拟量与光源离散量的选取原则。     

Windowing Design Method for Polynomial-Based Interpolation Filters

An efficient implementation for finding digitally the interpolated samples is the Farrow structure. It mimics digitally a hybrid system where a continuous-time (CT) signal is reconstructed using an analog reconstruction filter having a piecewise-polynomial impulse response. The interpolated samples are obtained by sampling reconstructed signal. This paper introduces a generalized design method for polynomial-based interpolation filters and Farrow structure. The proposed method also can be used to calculate the coefficients of Selva interpolator. In this approach, the ideal CT impulse response is truncated by using CT window functions. The obtained windowed impulse response is then approximated using the piecewise Taylor polynomial approximation. Length of the impulse response and degree of the approximating polynomial can be arbitrarily selected, and in this way the transition band width can be controlled. However, if CT fixed-window functions are used, the stopband attenuation is determined by window type and remains approximately constant with increase of length and order of the impulse response. The stopband attenuation can be controlled by using CT dynamic windows such as Kaiser window. The presented windowing design method is an effective tool for calculation of the Farrow structure coefficients, with filter performance that is comparable to the frequency domain design.     

A Diffraction Transfer Function Approach to the Calculation of the Transient Field of Acoustic Radiators

A computationally-efficient approach to the calculation of the transient field of an acoustic radiator was developed. With this approach, a planar or curved source, radiating either continuous or pulsed waves, is divided into a finite number of shifted and/or rotated versions of an incremental source such that the Fraunhofer approximation holds at each field point. The acoustic field from the incremental source is given by a 2-D spatial Fourier transform. The diffraction transfer function of the entire source can be expressed as a sum of Fraunhofer diffraction pattern of the incremental sources with the appropriate coordinate transformations for the particular geometry of the radiator. For a given spectrum of radiator velocity, the transient field can be computed directly in the frequency domain using the diffraction transfer function. To determine the accuracy of the proposed approach, the impulse response was derived using the inverse Fourier transform. The results obtained agree well with published data obtained using the impulse response approach. The computational efficiency of the proposed method compares favorably to those of the point source method and the impulse response approach.     

Formulation of time-domain algorithm for fiber Bragg grating simulation and reconstruction

A time-domain algorithm for fiber Bragg grating simulation is formulated. It provides a time-domain equivalent to the well-known spectral-domain transfer matrix algorithm for fiber Bragg grating simulation, and allows much more accurate impulse response to be simulated directly from a known fiber Bragg grating. Two different time-domain grating reconstruction algorithms are formulated, which are much more accurate and efficient than their spectral-domain equivalents due to their intrinsic infinite spectral representation and truly one-dimensional nature. Error studies are also conducted for the time-domain grating reconstruction algorithms. The high accuracy and efficiency of the new time-domain algorithms are well demonstrated with several simulation, reconstruction, and design examples.     

Least square identification of alias components of linear periodically time-varying systems and optimal training signal design

A least-squares (LS) method for identifying alias components of discrete linear periodically time-varying (LPTV) systems is proposed.The authors apply a periodic input signal to a finite impulse response (FIR)--LPTV system and measure the noise-contaminated output.The output of this LPTV system has the same period as the input when the period of the input signal is amultiple of the period of the LPTV system.The authors show that the input and the output can be related by using the discrete Fourier transform. In the frequency domain, an LS method can be used to identify the alias components. A lower bound on the mean square error (MSE) of the estimated alias components is given for FIR--LPTV systems.The optimal training signal achieving this lower MSE bound is designed subsequently. The algorithm is extended to the identification of infinite impulse response (IIR)--LPTV systems as well. Simulation results show the accuracy of the estimation and the efficiency of the optimal training signal design.     

Iterative site-based modeling for wireless infrared channels

We describe an iterative site-based method for estimating the impulse response of wireless infrared channels. The method can efficiently account for multiple reflections of any order. A simple geometrical model of indoor environments is presented which includes interior features such as partitions, people, and furniture, thus permitting accurate evaluation of shadowing effects. For a reflection order of three, the iterative method is over 90 times faster than the existing recursive technique. A computer implementation is described and used to demonstrate the efficiency and accuracy of the method     

Design and hybrid realization of FIR Nyquist filters with quantized coefficients and low sensitivity to timing jitter

A design of finite impulse response (FIR) Nyquist filters with zero intersymbol interference (ISI) and low sensitivity to timing jitter is presented. Using an affine scaling linear programming algorithm, a near-optimum quantized coefficient set can be obtained in a feasible computational time. By varying a parameter, the design provides a tradeoff between the tail energy of the impulse response in the time domain and the stopband of the magnitude response in the frequency domain. We also present a pipelined multiplier-free FIR filter realization with periodically time-varying (PTV) coefficients based on a hybrid form. The realizations exploit the coefficient symmetry to reduce the hardware by about one half. By placing most of the shifts followed by addition toward the back end of the structure, hardware is reduced due to the shorter wordlength of the adders. The proposed structure has a provision to increase the speed by adjusting a design parameter but at the expense of more hardware.     

基于联合对角化的声信号深度卷积混合盲分离方法

声信号在空间中的传播具有较强的多径效应,在接收端往往以卷积形式相互叠加,尤其在海洋、剧场等强混响条件下,混合滤波器冲激响应的长度会显著增加,现有的频域卷积盲分离算法将失效。为了消除长脉冲响应导致解混合模型失效的问题,该文对观测信号进行两次短时傅里叶变换(STFT),第1次STFT缩短了脉冲响应长度,第2次STFT将信号模型转化为瞬时盲分离,最终利用联合对角化(JD)技术估计出分离矩阵。与现有方法相比,所提方法解决了深度卷积混合下模型失效的问题,并且当源信号数较多或存在加性噪声时,可以得到更好的分离性能。仿真结果验证了方法的有效性和性能优势。