基于镜像源方法的室内声场脉冲响应仿真

针对基于室内声场脉冲响应描述声场音质特性时其精确度不高的问题,以经典镜像源方法为核心,讨论了基于变换频域和反射系数的改进算法.以矩形空间为具体实例,给定接收点和声源位置,并利用计算机仿真室内声场脉冲响应.在不同参数条件下,对各个算法进行了分析,估计了能量衰减曲线和房间混响时间.结果表明,基于变换频域和反射系数的改进算法具有更高的准确性和可靠性.     

基于传声器阵列的机器人声源定位系统

随着计算机技术和信号处理技术的发展,各种具有特殊功能的机器人应用到人们的生产生活中。听觉系统可以弥补机器人视觉系统存在的不足,使机器人在黑暗或有障碍物遮挡的环境中,利用声音信息感知环境中存在的目标,并对其定位,辅助机器人工作,尤其在灾难现场搜救、军事等领域,声源定位系统具有独特的优势。针对机器人声源定位系统进行研究,设计了基于传声器阵列的机器人声源定位系统,采用平面四元传声阵列,利用广义互相关算法,实现对声源的定位,具有较高的实用价值。     

Study of Sound Environment Influenced by the Crowd in Waiting Areas in General Hospitals

In this study, the crowd has been investigated and analyzed in waiting areas in large general hospitals in China in order to find the rules the variations of sound environment with the change of crowd. The field investigation, questionnaire, field-testing and computer simulation have been adopted. The results show that: the social/demographic characteristics of staff and patients are not significantly related to the satisfaction evaluation of sound environment; there is a significant correlation between the population density and LAeq of the background noise in waiting areas; when population density is 0, the LAeq of background noise is not 0 in waiting areas; the loudspeaker should be set in the waiting areas. Loudspeaker arrangements should be integrated into the ceiling lamp or construct facilities along the depth direction of the layout, and the two adjacent speakers recommended distance should be controlled at about 4 m. If the population density is controlled in the reasonable range, and sound absorption, noise reduction processing and electronic queuing system are adopted, sound environment of waiting areas will be built with noise interference relatively small in different population densities.     

《中德四季晨昏杂咏》（之八）的译文鉴赏

在诗歌翻译中,译文会因为译者的不同在风格上表现出差异.然而,译文的质量不外乎经受形美、音美和意美的考量.译者应从本国读者的文化背景上结合时代的脉搏来进行“再创造”,做到形式、音律和意蕴的完美统一.歌德的诗歌《中德四季晨昏杂咏》（之八）的四种中文译本各有特色,均在不同程度上实现了“三美”的融合.     

刀具磨损声谱特征的分析

刀具在切削过程中，会产生大量的声信号。文中运用统计方法对切削声信号进行了深入分析，并对可听阈内的声信号按低频带、中频带和高频带分别进行了功率谱分析。研究结果表明，切削声信号中100—8kHz频带内的特征频率分量反映了刀具磨损状态及其变化规律，这为利用声信号进行刀具实时在线监测奠定了基础。     

Digital sound broadcasting: Satellites or Terrestrial Networks?

When European laboratories decided to develop a digital sound broadcasting system (DSB), they specified three main conditions to fulfil:

• quality improvement up to the level of ‘CD’ sound, even in difficult reception conditions (mobile vehicles, etc)
• additional significant digital data transmissions in order to transform sound broadcasting into a really new service
• the possibility of a common system for satellite and terrestrial transmissions.

It is on these bases that the European project ‘Eureka 147’ defined the system called DAB (digital audio broadcasting). In 1992, the ITU Conference WARC 92 allocated 40 MHz to DSB in the L-band in the configuration of complementary terrestrial and satellite networks; nevertheless, the present state of technical possibilities makes such mixed networks almost unfeasible and the lack of available spectrum in VHF bands led a significant number of countries to envisage L-band for T-DAB. The situation could turn to a competition between terrestrial and satellite networks, especially because the bandwidth in L-band is not that large! France belongs to the countries facing this problem. L-band alone is intended to be used by T-DAB, and broadcasters taking part in the work of ‘Club DAB’ estimated that 20 MHz would be a minimum to ensure the success of T-DAB introduction. It is half of the DSB band. Splitting this band into two parts has already been decided by CEPT, but in the proportion of 1/3 for T-DAB. This organization intends to arrange a European planning meeting for T-DAB in July 1995, and, shortly after, several countries are ready to start the implementation of the terrestrial networks. At the same time, international broadcasters wonder whether satellite transmission could present an alternative to HF.     

混响噪声下声源定位与跟踪的多特征自适应IMM粒子滤波算法EI北大核心CSCD

针对混响噪声下声源定位精度低和鲁棒性弱等问题,提出了多特征自适应IMM粒子滤波算法.该算法以麦克风接收信号的多特征作为观测信息,采用空时相关和迭代滤波建立了时延选择机制和波束输出能量优化机制,并在两者的基础上构建了似然函数以获得合理的声源位置信息.考虑到说话人运动的随机性,给出了自适应IMM算法,通过在线粒子集生成并将不同过程方差的模型进行交互来拟合说话人的不同运动模式,改善了说话人跟踪系统的稳健性.仿真和实测结果表明,所提算法利用了多特征定位信息的互补性,降低了观测误差不确定性对声源位置估计的影响,增强了随机运动声源跟踪系统的鲁棒性,提高了系统的定位精度.     

负梯度声速下特征声线与传播时间快速求解方法

求解特征声线最直接的方法是采用"扫描-插值-迭代"的声线跟踪法,过程较复杂,计算速度较慢。将负梯度声速环境下特征声线的起始掠射角表示为声速、海水深度、声源与接收点相对位置的方程,通过采用量子粒子群算法求解方程直接获得掠射角,进而确定特征声线和传播时间。与声线跟踪法相比,所提出的方法由于不存在数值累计误差和角度插值误差,因此精度更高,另外速度也更快,适合浅海负梯度环境下特征声线与传播时间的快速求解。     

厅堂音质仿真设计中的声场模拟方法

音质仿真是模拟厅堂的听音效果,能给人一种身临其境的感受．本文提出了用微机再现室内脉冲响应,从而获得音质效果主观感受的一种方法,为评价音质效果提供一种手段．     

基于单片机的新型智能声级计的开发与研究

针对传统声级计采用纯硬件计权网络来实现噪声测量存在的操作复杂、检测精度低等问题,通过采用单片机作为智能声级计核心处理芯片,实现噪声的实时智能检测。阐述了智能声级计具体结构框架,并对系统的组成进行详细介绍,最后对声级计进行实验测试。