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。     

杭州剧院的声学模型试验

杭州剧院是一座多功能厅堂,观众厅的体积约10000m~3,能容纳2000观众席,已于1978年9月竣工使用,观众反映音质效果比较好。在我们进行音质设计过程中,为了及早对设计方案作出评价,作了1:10的声学模型试验。本文概括给出某些试验的结果,并与剧院建成后实测进行比较。模型试验包括(1)木条子墙和“船形”扩散体天花的声学特性,(2)吸声材料的模拟,(3)混响时间,(4)反射声,(5)声场分布,(6)观众对前次反射声的影响,(7)方向性扩散,(8)清晰度,(9)主观评价。模型和实物的对比结果:满场的混响时间除250Hz的差别较大外(约20％),其余频率比较接近(差别少于10％);反射声图形是相似的;声场分布的差别少于2dB;观众对从台口侧墙来的前次反射声有影响,衰减约6dB;方向性扩散低于实测值,这是由于制造模型时省略一些细部所引起;清晰度和主观评价与观众的实际感受大致相符。     

强背景下混响时间的非线性滤波M序列相关测量

本文提出采用非线性滤波抑制在强背景噪声环境中用Ｍ－序列相关法得到的房间脉冲响应中的残余噪声影响，以扩大混响衰减曲线的动态范围，从而达到能够在强背景噪声环境下准确测量混响时间的目的。首先讨论了影响Ｍ序列相关法测量混响时间的几个因素。其次，采用非线性滤波进一步抑制背景噪声的影响。结果表明，非线性滤波的效果相当显著。本文还在非白噪声背景条件下用该法进行强背景下的混响时间测量，结果与传统测量结果符合得很好     

基于统计的语音传输指数预测方法实验对比*

良好的言语可懂度是语言声厅堂声环境设计的重要目标,语音传输指数(STI)是言语可懂度的客观评价参量。在设计阶段就能对STI做出准确预测,对语言声厅堂的声环境控制具有重要意义。基于统计的STI预测方法是IEC 60268-16标准推荐的主要预测方法,但使用该方法时的影响因素以及预测精度,当前仍缺少系统性的实验对比研究。本文使用4个房间中13个测点共52种听音条件下的STI实测结果,分析了基于统计的STI预测方法的精度以及影响因素,结果表明：(1)考虑和不考虑直达声的两种预测方法,都存在着系统偏差,预测结果普遍小于实测值;在距离声源较近测点的预测都产生了较大误差,在距离声源较远测点的预测误差相对较小;(2)基于单指数衰变声场不考虑直达声的预测方法操作简单,但误差较大,52种听音条件的平均误差达到了-0.032,最大误差达到了-0.131;(3)考虑直达声的预测方法误差相对较小,52种听音条件的STI平均误差为-0.018,最大误差为-0.080;(4)预测精度主要受所使用的信噪比、扬声器的指向性、房间的声场条件等因素影响,并对这些因素的影响进行了对比分析。结果可为STI预测方法的使用者提供参考。     

南海北部海域声速剖面水平变化环境声学层析

为获取水平变化的海水声速分布,在等效平均海水声速反演基础上,提出了一种声速剖面水平变化环境声学层析方法。首先利用经验正交函数结合匹配场处理方法反演不同距离下的等效平均声速剖面,进而通过逐步迭代重构出随水平距离变化的声速场分布,并通过数值仿真及海上实验两方面验证了声速剖面水平变化环境声学层析方法的可行性及其准确性。结果表明:声速剖面水平变化环境声学层析结果与实验测量声速剖面均方根误差小于0.8 m/s,利用声传播信号基本能较为准确地重构出随距离水平变化的海水声速分布,为进一步层析中尺度三维结构及声呐水声环境保障应用奠定了基础。      

不同亚格子模型的对比分析及其运用

本文基于大涡模拟方法,对比讨论了四种不同亚格子模型的计算原理和仿真特性,并对等离子体激励下的凸包流场进行了研究,结果表明:WALE模型和KET模型的仿真结果比较准确;Smagorinsky-Lilly模型耗散偏大且模型系数不具普适性,仿真准确性较差;WMLES模型存在严重的过度耗散,各流动参数的仿真结果误差很大,无法准确捕捉流动转捩的发生;等离子体激励通过向附面层流体注入能量产生等效的壁面射流,可以显著延缓凸包流动分离并促进流动再附、抑制涡的发生,从而起到降低耗散,湍流减阻的作用。     

一种基于最大似然的混响时间盲估计方法*

混响是室内声学中的重要现象,在室内设计与音频信号处理中都需要测量或估计混响时间。本文改进了一种基于最大似然估计的混响时间盲估计方法,即采用说话人在房间中自然说话时发出的混响语音信号来估计混响时间的方法。该方法首先确定语音衰减段的最优边界,其次计算该衰减段的两个额外参数,据此筛选出符合条件的语音段,最后将满足条件的语音段采用最大似然估计得到混响时间估计值。在五个不同混响时间条件下的仿真表明,与已有方法相比,改进方法估计的混响时间同真实混响时间的偏差更小,方差更低,估计准确性较高。     

提高复杂燃烧流场羟基示踪测速精度的方法

为了提高流速测量精度,研究OH荧光图像背景抑制的方法.分析识别复杂燃烧流场中存在的背景干扰,构建染噪的数值仿真模型;基于干扰图像和OH标记线信号图像的特性,采用空间变换思想,提出了自适应差分法消除发动机燃烧室剧烈反应区域的燃烧OH荧光干扰;利用空域上信号与背景残余的差别,采用空间滤波法优化背景抑制结果,最后对仿真模型和实验图片对比,验证了该方法的有效性,处理前后峰值信噪比提高了11.83dB,信噪比提高了8.66dB,速度计算误差改善到了1.2%.该方法可有效的抑制背景噪声,提高测速精度,满足激光诊断系统对测量精度的要求.     

厅堂声学测量中不同激励声源的比较

基於脉冲响应积分的音乐厅和剧院观众厅声学特性的测量目前有三种使用不同激励声源的测试方法:人工脉冲声源、伪随机噪声序列(MLS)、以及用正弦扫频信号。这些技术各有其优缺点,在实际应用中为了方便根据具体情况选择不同的激励声源,通过在一个音乐厅现场的实测数据比较丁三种声源的实测结果,发现对混响时间测量三种不同的激励声源给出的结果基本一致,但是对明晰度和一些其他的指标,脉冲声源给出的结果与用MLS和扫频信号给出的结果有较明显的差别。对实际中如何选择具体的技术也做了建议。     

穆勒矩阵成像椭偏仪误差源的简化分析方法

针对穆勒矩阵成像椭偏仪的系统误差源提出一种简化分析方法,将光强曲线的理想傅里叶级数系数组与实际系数组进行近似匹配,建立穆勒矩阵测量误差与误差源参数之间的线性模型。针对解析式复杂的随机方位角误差,从统计学角度提出了一种等效噪声模型以分析其对测量结果的影响。采用上述简化方法系统分析了椭偏仪的6种系统误差源和2种随机误差源对穆勒矩阵测量结果的影响,并以一个典型光刻投影物镜的穆勒光瞳为检测对象,进行了检测仿真。仿真结果验证了所提方法分析的准确性。     

Feasibility of subjective speech intelligibility assessment based on auralization

Subjective speech intelligibility can be assessed by speech recorded in an anechoic chamber and then convolved with room impulse responses that can be created by acoustic simulation. The speech intelligibility (SI) assessment based on auralization was validated in three rooms. The articulation scores obtained from simulated sound field were compared with the ones from measured sound field and from direct listening in rooms. Results show that the speech intelligibility prediction based on auralization technique with simulated binaural room impulse responses (BRIRs) is in agreement with reality and results from measured BRIRs. When this technique is used with simulated and measured monaural room impulse responses (MRIRs), the predicted results underestimate the reality. It has been shown that auralization technique with simulated BRIRs is capable of assessing subjective speech intelligibility of listening positions in the room.     

Performance of an adaptive beamforming noise reduction scheme for hearing aid applications. II. Experimental verification of the predictions

A method to predict the amount of noise reduction which can be achieved using a two-microphone adaptive beamforming noise reduction system for hearing aids [J. Acoust. Soc. Am. 109, 1123 (2001)] is verified experimentally. 34 experiments are performed in real environments and 58 in simulated environments and the results are compared to the predictions. In all experiments, one noise source and one target signal source are present. Starting from a setting in a moderately reverberant room (reverberation time 0.42 s, volume 34 m3, distance between listener and either sound source 1 m, length of the adaptive filter 25 ms), eight different parameters of the acoustical environment and three different design parameters of the adaptive beamformer were systematically varied. For those experiments, in which the direct-to-reverberant ratios of the noise signal is +3 dB or less, the difference between the predicted and the measured improvement in signal-to-noise ratio (SNR) is -0.21+/-0.59 dB for real environments and -0.25+/-0.51 dB for simulated environments (average +/- standard deviation). At higher direct-to-reverberant ratios, SNR improvement is systematically underestimated by up to 5.34 dB. The parameters with the greatest influence on the performance of the adaptive beamformer have been found to be the direct-to-reverberant ratio of the noise source, the reverberation time of the acoustic environment, and the length of the adaptive filter.     

Sound field of typical single-bed hospital wards

The aim of this paper is to investigate the basic characteristics of sound fields in the increasingly concerned single-bed hospital wards, the effects of commonly used furniture, and the feasibility and strategic use of acoustic simulation techniques for such spaces. Two typical single-bed wards at a Critical Care Unit were selected as the case study site. Eight different room acoustic conditions were created by gradually moving pieces of furniture out of the ward. It has been shown that in the ward without acoustic ceiling, under empty room condition the RT ranged from 0.8 s, typically at low frequency, to 2.3 s at 630 Hz; the variations in both SPL and RT at different receivers were insignificant except at low frequencies, so that in such single-bed wards a diffuse field could be assumed; and longer RT and higher SPL were found when furniture was gradually moved out of the ward. In the ward with acoustic ceiling, the effect of furniture was less. There was a good agreement between measured and simulated RT and SPL, showing the usefulness of computer simulation for this kind of spaces. In the simulation, the effect of surface diffusion coefficient was generally insignificant, whereas the effect of edge diffusion should be taken into account, especially for the RT prediction. The feasibility of geometric simplification was also demonstrated.     

Development of an optimised, standard-compliant procedure to calculate sound transmission loss: design of transmission rooms

A numerical procedure to estimate the transmission loss of sound insulating structures is proposed based upon the technology of acoustic measurements and standards. A virtual laboratory (VL), namely, a numerical representation of a real laboratory consisting of two reverberation rooms meeting certain sound field quality criteria is designed. VL is to be used for the numerical simulation of standardised measurements under predefined, controlled, acoustic conditions. In this paper, the design and optimisation of VL is investigated. The geometry of the transmission rooms is designed following first principles, in order for diffuse field conditions and sufficiently smooth primary mode distribution in the low frequency to be achieved. A finite element-based optimisation procedure, introduced by the author in previous work, is extended to arbitrarily shaped rooms. It is used to predict the appropriate local geometric modifications so as for improved mode distribution and smoother sound pressure fluctuations of the transmission rooms in the low-frequency range to be achieved and low-frequency measurement reproducibility and accuracy to be increased. Steady-state acoustic response analysis is performed in order to quantify the acoustic field quality of the virtual transmission rooms in the frequency range of measurements. A method to calculate the total absorption, A, of the receiving room is introduced by simulation of the reverberation time measurement procedure using Transient acoustic response analysis. The acoustic performance of VL is overall considered and is shown to meet in a sufficient degree, relative laboratory measurement standards in the frequency range of 100÷704 Hz.     

Accuracy of speech transmission index predictions based on the reverberation time and signal-to-noise ratio

This paper examines the accuracy of the speech transmission index (STI) calculated from the reverberation time (T) and signal-to-noise ratio (L_SN) of enclosed spaces. Differences between measured and predicted STIs have been analysed in two rooms (reverberant vs. absorbent), for a wide range of absorption conditions and signal-to-noise ratios (sixteen tests). The STI was measured using maximum length sequence analysis and predictions were calculated using either measured or predicted values of T and L_SN, the latter assuming diffuse sound field conditions. The results obtained for all the conditions tested showed that STI predictions based on T and L_SN tend to underestimate the STI, with differences between measured and predicted STIs always lower than 0.1 (on a 0.0–1.0 scale), and on average lower than 0.06. According to previous research, these differences are noticeable and therefore non-negligible, as 0.03 is the just noticeable difference in STI. The use of either measured or predicted values of T and L_SN provided similar STI predictions (i.e. non-noticeable changes), with differences between predictions that are on average lower than 0.03 for the absorbent room, and lower than 0.01 for the reverberant room.     

Fast estimation of speech transmission index using the reverberation time

The paper presents the function of STI in the domain of reverberation time. Through the application of the said function, we can quickly estimate the speech transmission index, knowing only the time of room reverberation. For that purpose we applied a known method which consists in physical estimation of speech intelligibility basing on the modulation transfer function (MTF) determined in a room. Then, the STI was described using a logarithmic function whereof argument was the room reverberation time. To verify the model, reverberation times of six rooms were measured. The selected rooms were very different deliberately. They had different cubature and shape. The selection included a small cuboid, lecture halls and a church. Then, the same rooms were modeled in the ODEON version 11.23 and their reverberation times were determined. Furthermore, the STI was determined in the ODEON and then compared with the reverberation time obtained in effect of fast estimation. The statistical verification with the use of correlation index and regression equation has demonstrated that the fast estimation yields results close to those obtained in the computer simulation in ODEON. We obtained the correlation index at the level close to 1. Furthermore, the test probability at the level lower than 0.05 bespeaks of a statistically significant linear relation for the confidence level of 95%.     

Virtual models for the prediction of acoustic fields of Manuel de Falla Auditorium in Granada,Spain

This paper presents the results of the acoustic simulation of the Manuel de Falla Auditorium (MFA), Granada, Spain, carried out in CATT-Acoustic and CATT–TUCT. The 3D geometrical acoustic model created for this purpose is tuned by adjusting the absorption and scattering coefficients of the convex reflector ceiling of the auditorium through an iterative process aimed at rendering differences between simulated and measured values for reverberation time in each octave frequency band lower than 5%. The model is validated by comparing simulated and measured figures for the most significant monaural and binaural room acoustic parameters assessed at multiple locations both on the stage and within the audience areas, then by performing both spectral and spatial analyses of their behaviour, and ultimately studying dissimilarities between the two sets of data in terms of just noticeable difference (JND). The influence of the degree of detail of the geometric model on the outcome is also considered by incorporating the 44 iconic lamps specifically designed for this enclosure, into the model. Likewise, the acoustic effect of an already executed seating replacement in the performance space is analyzed in relation to the degree of occupancy of the room. The simulated results confirm the appropriateness of the choice of the new seating for the refurbishment of the concert hall.     

Wayside acoustic defective bearing detection based on improved Dopplerlet transform and Doppler transient matching

The diagnosis of train bearing defects plays a significant role in maintaining the safety of railway transport. However, the phenomenon of Doppler Effect in the acoustic signal recorded by the wayside Acoustic Defective Bearing Detector (ADBD) system leads to the difficulty for fault diagnosis of train bearings with a high moving speed. This paper proposes a double-searching solution based on improved Dopplerlet transform and Doppler transient matching to overcome the difficulty in wayside acoustic bearing diagnosis. In the solution, the first searching procedure is to extract necessary parameters of Doppler Effect under the situation with very low signal-to-noise ratio (SNR) based on an improved Dopplerlet transform. Using the obtained parameters, the Doppler Effect can be embedded into the constructed periodic Laplace wavelet transient models. Subsequently, the second searching procedure is conducted to search fault impact period of the defective bearing through an operation, called Doppler transient matching, which is to calculate the correlation coefficient between the Doppler transient model and the filtered raw signal with the Doppler Effect. The proposed double-searching algorithm can adapt to the real Doppler Effect situation and extract the exact fault impact period from the Doppler distorted signal, and thus shows powerful capability to analyze wayside acoustic signals from train bearings. The proposed wayside acoustic diagnostic scheme is verified by means of a simulated Doppler distorted signal with a very low SNR (−20 dB) and the experiments conducted on train bearings. The results indicate that the proposed algorithm is effective and has obvious advantages for ADBD system.     

Prediction of speech intelligibility for public address systems in traffic tunnels

Traffic tunnels are generally hostile acoustic environments, both in terms of reverberation and ambient noise levels. Public address (PA) systems used to convey spoken warnings must meet stringent design requirements in order to produce sufficiently intelligible speech. To be able to predict PA system performance at tunnel design time, two different speech transmission index (STI) calculation procedures were implemented. The first procedure predicts the STI based on ray-tracing simulations. Comparison with measured STI data showed that this simulation approach yields accurate intelligibility estimates. However, the procedure is time-consuming and too complex to be used by non-specialists. For this reason, a second (simpler and more efficient) procedure was developed, based on fixed non-linear regression, statistically deriving prediction functions from measured data and ray-tracing results. This procedure was compared to the approach based on ray tracing, and found to yield STI predictions closely matching those of the ray-tracing model.     

Effects of unattended speech on performance and subjective distraction: The role of acoustic design in open-plan offices

Unattended background speech is a known source of cognitive and subjective distraction in open-plan offices. This study investigated whether the deleterious effects of background speech can be affected by room acoustic design that decreases speech intelligibility, as measured by the Speech Transmission Index (STI). The experiment was conducted in an open-plan office laboratory (84 m²) in which four acoustic conditions were physically built. Three conditions contained background speech. A quiet condition was included for comparison. The speech conditions differed in terms of the degree of absorption, screen height, desk isolation, and the level of masking sound. The speech sounds simulated an environment where phone conversations are heard from different locations varying in distance. Ninety-eight volunteers were tested. The presence of background speech had detrimental effects on the subjective perceptions of noise effects and on cognitive performance in short-term memory and working memory tasks. These effects were not attenuated nor amplified within a three-hour working period. The reduction of the STI by room acoustic means decreased subjective disturbance, whereas the effects on cognitive performance were somewhat smaller than expected. The effects of room acoustic design on subjective distraction were stronger among noise-sensitive subjects, suggesting that they benefited more from acoustic improvements than non-sensitive subjects. The results imply that reducing the STI is beneficial for performance and acoustic satisfaction especially regarding speech coming from more distant desks. However, acoustic design does not sufficiently decrease the distraction caused by speech from adjacent desks.