室内脉冲声场中声强的测量研究

采用多次重复发射的声源配合装置在缓慢转动平台上的接收传声器所组成的测量系统(简称为RRS),它等效于多个完全匹配的传声器组成的圆型阵列。文中着重阐述了借助此技术测量房间内脉冲声场中声强的时间与空间分布特性。阐明了以RRS测量入射脉冲声强的理论基础,分别对离散的脉冲声场及扩散声场提出了测量声强的定量关系式,对实验结果及应用作了分析与讨论。     

A hybrid MLS technique for room impulse response estimation

The measurement of room impulse response (RIR) when there are high background noise levels frequently means one must deal with very low signal-to-noise ratios (SNR). If such is the case, the measurement might yield unreliable results, even when synchronous averaging techniques are used. Furthermore, if there are non-linearities in the apparatus or system time variances, the final SNR can be severely degraded. The test signals used in RIR measurement are often disturbed by non-stationary ambient noise components. A novel approach based on the energy analysis of ambient noise - both in the time and in frequency - was considered. A modified maximum length sequence (MLS) measurement technique, referred to herein as the hybrid MLS technique, was developed for use in room acoustics. The technique consists of reducing the noise energy of the captured sequences before applying the averaging technique in order to improve the overall SNRs and frequency response accuracy. Experiments were conducted under real conditions with different types of underlying ambient noises. Results are shown and discussed. Advantages and disadvantages of the hybrid MLS technique over standard MLS technique are evaluated and discussed. Our findings show that the new technique leads to a significant increase in the overall SNR.     

Relationship between impulse response and other types of room acoustical responses

This paper presents a method of calculating sound build up, steady state and sound reduction phenomena from the impulse response of rooms. The noise components of both the testing signal and the room response are omitted and wave phenomena occurring in the room are also neglected. A situation corresponding to the geometrical propagation of sound is thus simulated. The resulting formulae are an extension of corresponding methods for the numerical modelling of acoustical fields in rooms. In this way, as well as the impulse response, sound build up and reverberation curves may also be obtained. An example using the ray tracing technique is presented.     

基于三元伪随机序列集的多声源房间脉冲响应测量方法*

在多输入多输出系统中,诸多情况下需并行测量任意输入输出之间的脉冲响应。虽然利用以Kasami序列集为代表的二元伪随机序列集良好的相关特性可以实现多声源并行测量,但是囿于其编解码信号相同且为二元信号的特点,使得其相关特性仍存在较大的可优化空间。该文提供了一种基于三元伪随机序列集的多声源房间脉冲响应测量方法,具有更好的相关特性,提高了测量准确度。仿真和实验验证了该测量方法的可行性和准确性。在封闭空间几何模型构建等领域中,可使用该方法提高测量效率并降低由于各通道不同步导致的误差。     

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.     

考虑界面声散射的室内声脉冲响应计算机仿真新算法

本文提出一种考虑界面声散射的室内声脉冲响应的计算机仿真新算法，该算法通过应用动态堆栈和虚拟内存，解决了模拟了中计算可能失运控制的问题，作为例子，文中对二个矩形房间的声脉冲响应进行了仿真。     

Exergy method for estimating the ramjet specific impulse

The exergy method is developed for computing the ramjet thrust-economic characteristics with regard for real thermodynamic properties of combustion products when using as fuel the hydrogen and hydrocarbon fuel for the freestream Mach numbers M = 4 ÷ 14. The estimates for the specific impulse of the given engine using the presented technique are shown to agree with the estimates computed by other authors. The computational method is intended for obtaining the ramjet characteristics and conduction of the parameter analysis at the research initial stage as well as for its use at the conceptual developments of hypersonic flying vehicles.     

房间脉冲响应模拟法及混响时间盲估计应用

在构建混响语声数据集时,由于缺乏真实长混响房间脉冲响应且模拟的房间脉冲响应与真实不符,因而导致数据驱动的混响时间盲估计模型性能下降。提出了一种基于条件生成对抗网络的房间脉冲响应模拟法,该方法利用真实的房间脉冲响应训练条件生成对抗网络,可以根据指定的混响时间模拟更加真实的房间脉冲响应。使用不同方法模拟的房间脉冲响应构建训练集用于训练盲估计模型,通过声学实验评估模型性能。实验结果表明,由该方法模拟的房间脉冲响应训练的估计模型在不同信噪比下均具有最小的均方根误差且在长混响情况下显著优于其他模型。     

Dynamic measurement of room impulse responses using a moving microphone

A technique for the recording of large sets of room impulse responses or head-related transfer functions is presented. The technique uses a microphone moving with constant speed. Given a setup (e.g., length of the room impulse response), a careful choice of the recording parameters (excitation signal, speed of movement) leads to the reconstruction of all impulse responses along the trajectory. In the case of a moving microphone along a circle, the maximal angular speed is given as a function of the length of the impulse response, its maximal temporal frequency, the speed of sound propagation, and the radius of the circle. As a result of the presented algorithm, head-related transfer functions sampled at 44.1 kHz can be measured at all angular positions along the horizontal plane in less than 1 s. The presented theory is compared with a real system implementation using a precision moving microphone holder. The practical setup is discussed together with its limitations.     

小阻尼界面房间声传输函数和声脉冲响应函数的有限元素法

有限元法可用于以声波动方程为基础通过数值计算求解室内声场,适用于分析界面阻抗非均匀分布和复杂形状房间内声场的低频特性。本文首先介绍了小阻尼界面条件下室内声场简正方式、声衰变系数、混响时间的ＦＥＭ计算方法。在此基础上导出了房间内两点之间声传输函数和声脉冲响应函数的ＦＥＭ计算模型,并以矩形房间为例详细讨论了有关细节。本文所讨论的计算模型可以映房间内不同的声源点,接收点位置上的声压频谱特性和脉冲响应的时     

利用时域波形快速反演海底单参数的方法

海底单参数反演由于反演维数少, 较之多维反演具有实施简单快捷的优点. 首先定义小掠射角下反射损失随掠射角变化率F_dB为单个参数描述海底性质, 构建单参数反演模型; 其次, 从简正波与射线理论出发, 对时域脉冲波形与海底反射性质之间的关系进行了推导, 并讨论衰减机制; 最后, 基于该关系设计出利用最小二乘法对时域波形进行反演的方法. 对北黄海实验数据进行了实际反演, 通过海底采样验证反演结果对海底性质估计的正确性, 同时基于反演结果的传播损失预报值与实测值符合良好. 该方法实施仅需要单个水听器, 待测物理量少, 所获结果较为可靠, 且能满足大多数应用的需要. 关键词： 单参数 地声反演 浅海 海底反射损失     

Reverberation time measurement by the product of two room impulse responses

The decay function for the evaluation of the reverberation time is often obtained by the method of the backward integration of a squared room impulse response as suggested by M.R. Schroeder more than four decades ago. Since then much work has been published about its implementation. However, soon after the initial exploitation of the method, it was realized that the effects of the background noise contaminating the room impulse response required a careful consideration for accomplishing better results.This paper describes an alternative method dealing with the problem of the backward integration of noisy room impulse responses. This method is based on the processing of two impulse responses sequentially recorded for a fixed source and receiver arrangement in a room. Statistical criteria are proposed to identify how the effect of the noise corrupts the level decay curve using a noise-free synthesized room impulse response as well as measurements performed in a real room.     

Localizing nearby sound sources in a classroom: binaural room impulse responses

Binaural room impulse responses (BRIRs) were measured in a classroom for sources at different azimuths and distances (up to 1 m) relative to a manikin located in four positions in a classroom. When the listener is far from all walls, reverberant energy distorts signal magnitude and phase independently at each frequency, altering monaural spectral cues, interaural phase differences, and interaural level differences. For the tested conditions, systematic distortion (comb-filtering) from an early intense reflection is only evident when a listener is very close to a wall, and then only in the ear facing the wall. Especially for a nearby source, interaural cues grow less reliable with increasing source laterality and monaural spectral cues are less reliable in the ear farther from the sound source. Reverberation reduces the magnitude of interaural level differences at all frequencies; however, the direct-sound interaural time difference can still be recovered from the BRIRs measured in these experiments. Results suggest that bias and variability in sound localization behavior may vary systematically with listener location in a room as well as source location relative to the listener, even for nearby sources where there is relatively little reverberant energy.     

On estimating system damping from frequency response bandwidths

As damping determines the maximum vibratory response of a system at resonance, reliable estimates of damping are critical both to the design and qualification of systems to be subjected to a vibratory environment and for the evaluation of the effectiveness of modifications or additions provided to increase damping. The sharpness of the frequency response at resonance is often used for this purpose, quantified by the width of the frequency range (bandwidth) for which the response is above some fraction of the maximum response. Several aspects of the use of bandwidth methods in interpreting test results are considered. It is shown that the use of an excessive rate of change of test frequency in a sine sweep leads to overestimates of system damping. A criterion is offered for the identification of the maximum sweep rate for which an observed frequency response function provides a true indication of system damping, rather than an erroneous value dominated by the sweep rate. Applicability of the criterion is demonstrated through the use of results from actual tests. Excessive sweep rates are shown to inflate estimates of system loss factors above the true values in proportion to the square root of the sweep rate. It is also demonstrated with a specific form for an amplitude-dependent stiffness that the resulting nonlinearity can lead to erroneous observations of bandwidth frequencies, as well as the need for further reductions in the sweep rate.     

Estimation of the impulse response of ultra-wide-band systems

The time and frequency algorithms of impulse response recovery in linear ultra-wide-band systems by a finite set of digital data presenting the input and output signals are proposed. The results of approbation of these algorithms by numerical simulation in the presence of additive Gaussian noise in the output signal are given. It is shown that the systems with noise-like signals at the input have the largest noise immunity in the estimation of impulse response. Institute for High-Current Electronics of the Russian Academy of Sciences, Tomsk, Russia. Translated from Izvestiya Vysshikh Uchebnykh Zavedenii, Radiofizika, Vol. 41, No. 9, pp. 1195–1206, September, 1998.     

Experimental determination of the impulse response of cascade arcsusing a correlation technique

Low-current cascade arcs are one of the simplest configurations for investigating the behaviour of low-temperature arcs. In cascade experiments, arc conditions may be strictly controlled, in which case the results become reproducible, allowing detailed measurements of high resolution to be made. The response of cascade arcs to a small change in the current gives insights into the physical processes taking place in an arc, such as ionization/recombination, diffusion and energy transfer between the electrons and the heavy particles. In this work, a new method of measuring time constants is described, in which correlation techniques utilising a pseudo-random binary sequence are employed to deduce the impulse response of the arc. This type of experiment provides a useful check on the validity of the collision models used for the reaction processes in the arc plasma and it is planned to investigate other gases in the future. In this paper, the experimental arrangement and the data analysis are discussed and experimental results are presented