电声音乐中的物理学问题--物理与音乐专题之六

一般而言，在电声乐器中，电声乐器与电子乐器不是一个概念．用传统的乐器发声，然后用电声放大的称为电声乐器，如电吉他、电扬琴、电子手风琴等．由电子振荡器产生声的称为电子乐器，如电子琴、电子钢琴等．数字式电子音响合成器也是电子乐     

在群论框架下电子三重态与声子耦合的理论研究

本文基于绝热近似和群论导出了电声耦合系统的哈密顿量的一般形式,讨论了电声耦合系统中的电子算符和活跃的声子模式. 应用幺正平移变换和能量最小化方法,进一步计算了正四面体群下T*(e+t₂)杨-泰勒系统中的激发态能量,从对称性的角度分析了T₁电子态的能级分裂以及晶格体系的对称性破缺,得出了对称性的破缺方式和电声耦合系统密切相关的结论. 结果表明:通过群论与对称性分析完全可以定性地解释由于电声耦合所造成的简并电子 关键词： 电声耦合 杨-泰勒畸变 活跃的杨-泰勒声子模式 电声耦合哈密顿量     

会议报道

1987年9月中国声学学会电声学分科学会、电子工业部电声情报网,武汉声学学会在武汉联合召开了学术交流会.大会共收到论文28篇,其中九篇集中讨论了电声器件与整机的接口技术,包括传声器与整机的配接.调音台与功率放大器的连接,民用整机中扬声器的配接等.论文中提出要在功率、灵敏度、电压、阻抗、接地等方面考虑匹配,并且介绍了扬声器系统的动     

电声学科的跨越特点和应用进展

本文综述了电声学科由于具有跨越科学与艺术两大领域和跨越多种学科的性质所带来的研究和应用特点,提出了几项近些年值得重视的应用领域和可能发展趋势,希望引起同仁讨论。鉴于我国在其中有些方面有突出成绩,本文将特别加以介绍,也希望得到大家的关心。本文还有一个附带的期望：促进非电声学科的声学工作者了解和关心电声学科的核心。     

第一性原理计算MgB2薄膜拉伸对超导电性的影响

应用全势线性响应线性糕模轨道方法计算MgB₂的电子能带结构、声子谱及电声子耦合常数,并讨论MgB₂的超导电性.通过比较MgB₂薄膜双轴拉伸前后超导电性的变化可以看出，随着a轴晶格常数增大和c轴晶格常数减小,声子谱中硼的E_2g声子频率显著下降，使得电声子耦合强度λ和声子对数平均频率ω_ln增强,提高了MgB₂薄 关键词： 超导电性 能带结构 声子频率 电声子耦合     

Nd∶YAG中电声耦合非Markovian过程的激发光谱

用Brown振子模型研究YAG基质材料中Nd^3+ 4f电子跃迁的电子-声子相互作用过程，详尽推导电子．声子耦合的表达式，计算了电声耦合常数不同时的激发光谱，结果表明，电声耦合作用使电子跃迁的吸收峰两边产生声子吸收峰，电声耦合作用较强时，可以产生多级声子吸收峰．通过模拟实验激发光谱，得到样品的电声耦合常数和声子频率等参数，和半导体比较，样品的电声耦合常数较小，说明4f电子．声子的耦合属弱相互作用．用这些参数计算时间分辨的荧光相干光谱，其结果与实验结果一致．     

2006年国际电声专家讲座在国内三地分别举办

为了促进高层次的国际电声技术交流，激发国内具有更多国际水平的电声研发，由中国电子学会声频工程分会和深圳意富音响公司联合主办，天津科技大学，天津轻工工程学会，天津市声学学会，国际电声网联合协办，2006国际电声大师中国行讲座于2006年9月4日、7日和12日分别在东莞、天津、嵊州等三地举行。这三场活动分别由嘉茂、真美、天乐赞助。     

功率换能器电声效率测量的初步研究

本文初步研究了利用值流量热法,在牛角尖形量热计中,对功率换能器在实用功率状态下电声效率的测量.这种测量方法避免了已往在小信号情况下测量功率换能器电声效率的一些不足之处.其中对恒流量热法测量原理、量热计形式和吸收液的选择等都作了简要的讨论,并给出了部分换能器的测量结果.     

《电声词典》

为适应电声科学技术的飞速发展形势,机械电子工业部电声专业情报网组织了有关电声专业的研究所、高等院校、设计院和工厂等有经验专家教授组成编委会,撰写了《电声词典》,聘马大猷教授任顾问,由沈(山豪)主编,国防工业出版社出版。     

杂质中心吸收宽带的电声子耦合问题(Ⅱ)

在这篇文章中,我们用一维晶格模型讨论了杂质中心中电声子耦合强度的问题。用场论方法严格地解得了含杂质晶格之运动方程的本征函数。由此得到了电声子耦合强度的解析表示式,它是用声子的波数k、表示相互作用范围的参量λ以及杂质参量P=γ′/γ解析地表示出来的。其中γ′和γ分别为杂质与近邻之间和一般近邻之间的力常数。对结果的分析表明,只改变质量的杂质不影响电声子耦合;导致力常数变化的杂质对电声子耦合有显著的影响。当有奇的局域模出现时,在离子晶体中它对带宽的贡献可以比带内模的贡献大很多。尤其是在离子晶体中有可能出现所谓“临界散射”,这时带内模的贡献可能变得很小,而主要的贡献几乎全来自于局域模。相反地,在非极化晶体中,局域模的贡献一般是很小的。文中最后讨论了由一维模型得到的结论对于三维晶体可能有的意义。     

Digital audio signal encryption with a virtual optics scheme

A novel approach for enciphering digital audio signal is presented in this article. The operating principle of the approach is on the basis of a virtual optics scheme whereby we apply both virtual wavelength and virtual diffraction distance in conjunction with a complex-valued random mask to design multiple-locks and multiple-keys in the course of audio data encryption and decryption. The audio signal is converted into a sound map, thus the encoded sound map can be encrypted as done for an image. Numerical experiment results show that proposed method is with high sensitivity to parameter discrepancy of virtual optical scheme. The correlation analysis of decrypted audio signal in comparison with original signal is also performed to verify the capability of presented virtual optics scheme for audio signal encryption.     

On the relative importance of spatial and timbral fidelities in judgments of degraded multichannel audio quality

Mean opinion score ratings of reproduced sound quality typically pool all contributing perceptual factors into a single rating of basic audio quality. In order to improve understanding of the trade-offs between selected sound quality degradations that might arise in systems for the delivery of high quality multichannel audio, it was necessary to evaluate the influence of timbral and spatial fidelity changes on basic audio quality grades. The relationship between listener ratings of degraded multichannel audio quality on one timbral and two spatial fidelity scales was exploited to predict basic audio quality ratings of the same material using a regression model. It was found that timbral fidelity ratings dominated but that spatial fidelity predicted a substantial proportion of the basic audio quality.     

Digital audio information hiding based on Arnold transformation and double random-phase encoding technique

In this paper, Arnold transformation and double random-phase encoding technique widely used in digital image information hiding are introduced to digital audio information hiding. The digital audio is transformed into a 2-D image called sound map and then the sound map will be divided into many windows and each window will be encrypted based on the Arnold transformation. Finally sound map will be re-encrypted based on double random-phase encoding technique. This method offers many advantages for digital audio information hiding: improve security and high attack immunity.     

自制声波干涉仪

用自制音频振荡器给2个电声性能相同的扬声器提供音频信号使其发声,在空间形成两列频率相同的声波,并通过声音的强弱来演示声波的干涉现象.     

Down-mixing of multi-channel audio for sound field reproduction based on spatial covariance

This article describes a method for automatic down-mixing multi-channel audio content on the basis of spatial covariance. Such a down-mixing method should be able to convert the signal of the original multi-channel audio system into that for an alternative system with the lesser number of channels, while maintaining the spatial impression of sound. Moreover, it should take into account the listener’s position and transfer function. Wave surface control and convolving the head related transfer function are techniques used in sound field control or reproduction. We consider that the spatial impressions of a sound field, which we perceive through our ears, are reproduced by preserving the relative relationship between observation points even if the wave surface is not completely controlled. Takahashi et al. proposed a new sound field reproduction method that we named “SOund field Reproduction based on sPAtial Covariance” (SORPAC). SORPAC can control the point-to-point covariance in a sound field. We expect that this sound field reproduction method based on spatial covariance can be applied to down-mixing of multi-channel content because SORPAC does not require the listener’s position or transfer function. This article describes SORPAC and its characteristics. We used SORPAC for down-mixing audio content. We confirmed that SORPAC-based down-mixing could accurately reproduce the interaural cross correlation (IACC) in relation to the listener’s position.     

立体声技术近年来的进展

本文总结和评论了立体声技术近年来的进展，包括声像定位理论和实验，传统的双通路立体声系统的改进，环绕产系统，高清晰度电视声系统，假头型立体声系统，数字声频技术在立体声的应用等．文中最后还叙述了对立体声今后发展的展望．     

调控明区声场重建性能和暗区声能量的综合控制方法

采用扬声器阵列的空间局部声场重建是指控制扬声器阵列输入的幅值和相位,使得目标声场聚焦于特定区域(明区)内,同时降低其他区域(暗区)的声能量。为满足车内乘客对独立视听环境需求的不断提高,研究车内局部声场重建。考虑到扬声器有驱动功率限制和传递函数存在测量误差,提出一种调控明区声场重建性能与暗区声能量平衡的鲁棒性控制方法(PSR-RC,Personal Sound Reproduction with Robust Control),采用路径跟踪法获得不同频率下的最佳权重因子。仿真结果与实车测试性能显示,PSR-RC算法保证明区重建声场性能的基础上,通过最小化暗区能量,获得了较好的明暗区声能量对比度。仿真与实车试验结果保持了较好的一致性,控制点区域的控制效果与非控制点区域的控制效果也保持了较好的一致性,验证了PSR-RC算法的正确性和可靠性。      

Sound field monitoring by tomographic electronic speckle pattern interferometry

Time-averaging electronic speckle pattern interferometry (ESPI) allows to record the phase modulation of light that has propagated through a sound field. When such data are collected from different projection directions the three-dimensional spatial distribution of amplitude and phase of the sound are obtained by tomographic back projection. The performance of such a setup increases with the number of projection directions, the number of effective resolution elements in the detector, and the number of recordings taken in averaging. These efforts, however, compete with the need for acceptable recording and processing times. Recent improvements in time-averaging ESPI enable even demanding applications in sound field monitoring. This is demonstrated in the design of a 38.5 kHz ultrasound source composed of a large number of individual piezoelectric transducer elements and intended to generate highly directive audio sound by nonlinear mixing in air (parametric array). The success of this method relies essentially on a non-intrusive control of the spatial homogeneity of the ultrasound field. Tomographic ESPI data have guided in a delicate alignment of the transducer elements yielding the expected narrowing of the angular radiation of the audio sound.     

基于LabVIEW和声卡的音频信号采集与分析系统设计

详细介绍了系统的开发背景,软件结构和特点,系统地分析了数据采集硬件和软件设计技术,在此基础上以声卡为数据采集卡,以LabVIEW为开发平台设计了音频信号采集与分析系统。这种系统具有界面友好,维护简单等优点。系统不但可以实现对音频信号进行采集,还可以对采集的信号进行分析和保存。可以根据需要改变系统的功能,具有很好的教学实...