Multichannel audio rendering using amplitude panning

Spatial audio is a field that investigates techniques to reproduce spatial attributes of sound (e.g., direction, distance, width of sound sources, and room envelopment) to the listener. Such attributes cannot be reproduced accurately with one loudspeaker, therefore two-channel stereophony was introduced, later extended to different surround formats having five to eight loudspeakers. An even more accurate reproduction of spatial attributes can be achieved with loudspeaker setups often found in theaters and in some public venues, which contain a large number of loudspeakers around listeners, possibly also above and/or below them.     

A Stochastic Approximation Method with Enhanced Robustness for Crosstalk Cancellation

The objective of acoustic crosstalk cancellation is to use loudspeakers to deliver prescribed binaural signals (that reproduce a particular auditory scene) to a listener's ears, which is useful for 3-D audio applications. In practice, the actual transfer function matrix will differ from the design matrix, because of either the listener's head movement or rotation, etc. Crosstalk cancellation system (CCS) is very non-robust to these perturbations. Generally, in order to improve the robustness of CCS, several pairs of loudspeakers are needed whose position varies continuously as frequency varies. With the help of assumed stochastic analysis, we propose a stochastic robust approximation crosstalk cancellation method based on random perturbation matrix modeling the variations of the transfer function matrix. Under the free-field condition, simulation results demonstrate the effectiveness of the proposed method.     

Surrounded by sound

The authors discuss immersive audio systems and the signal processing issues that pertain to the acquisition and subsequent rendering of 3D sound fields over loudspeakers. On the acquisition side, recent advances in statistical methods for achieving acoustical arrays in audio applications are reviewed. Classical array signal processing addresses two major aspects of spatial filtering, namely localization of a signal of interest, and adaptation of the spatial response of an array of sensors to achieve steering in a given direction. The achieved spatial focusing in the direction of interest makes array signal processing a necessary component in immersive sound acquisition systems. On the rendering side, 3D audio signal processing methods are described that allow rendering of virtual sources around the listener using only two loudspeakers. Finally, the authors discuss the commercial implications of audio DSP     

Adaptive inverse filters for stereophonic sound reproduction

A general theoretical basis for the design of adaptive digital filters used for the equalization of the response of multichannel sound reproduction systems is described. The approach is applied to the two-channel case and then extended to deal with arbitrary numbers of channels. The intention is to equalize not only the response of the loudspeakers and the listening room but also the crosstalk transmission from right loudspeaker to left ear and vice versa. The formulation is a generalization of the Atal-Schroeder crosstalk canceler. However, the use of a least-squares approach to the digital filter design and of appropriate modeling delays potentially allows the effective equalization of nonminimum phase components in the transmission path. A stochastic gradient algorithm which facilitates the adaptation of the digital filters to the optimal solution, thereby providing the possibility of designing the filters in situ, is presented. Some experimental results for the two-channel case are given     

控制声像距离的二维FIR滤波器设计

声像距离控制是实现三维视听系统的关键技术之一。现有的音频环绕声技术产生的声像距离固定,且扬声器环形结构不利于与视听系统融合。针对上述问题,本文提出一种利用线型扬声器阵列通过二维FIR滤波器控制声像距离的方法。首先根据声波的传播特点,通过调整声源到波前的时间延迟,改变波前半径产生了虚拟声像。将时间延迟作为二维FIR滤波器的群延迟,使滤波器具有线性相位。考虑到宽带信号,设计具有楔形过渡带的滤波器幅频响应,使在期望宽度内形成波束。最后给出了滤波器实现步骤以及仿真结果。实验表明利用二维FIR滤波器在线型扬声器阵列中实现了声像距离的控制,输出稳定的波束。该方法改变了声像的距离,增强了沉浸感,线型扬声器更容易与视听系统相结合。      

A Spatial Audio System Using Multiple Microphones on a Rigid Sphere

The main purpose of a spatial audio system is to give a listener the same impression as if he/she were present in a recorded environment. A dummy head microphone is generally used for such purposes. Because of its human‐like shape, we can obtain good spatial sound images. However, its shape is a restriction on its public use and it is difficult to convert a 2‐channel recording into multi‐channel signals for an efficient rendering over a multi‐speaker arrangement. In order to solve the problems mentioned above, a spatial audio system is proposed that uses multiple microphones on a rigid sphere. The system has five microphones placed on special points of the rigid sphere, and it generates audio signals for headphone, stereo, stereo dipole, 4‐channel, and 5‐channel reproduction environments. Subjective localization experiments show that front/back confusion, which is a common limitation of spatial audio systems using the dummy head microphone, can be reduced dramatically in 4‐channel and 5‐channel reproduction environments and can be reduced slightly in a headphone reproduction     

Subjective Listening Experiments on a Front and Rear Array‐Based WFS System

Wave field synthesis (WFS) has been gathering more and more attention recently due to its ability to perfectly reproduce an original sound field. However, to realize theoretically perfect WFS, a four‐sided loudspeaker array that encloses the listener is required. However, it is difficult to build such a system except in large listening spaces, such as a theater or concert hall. In other words, if the listening space is a home, installing a side loudspeaker array is impractical. If the two side walls located to the left and right of the listener can be omitted, a setup using only front and rear loudspeaker arrays may be a solution. In this letter, we present a subjective listening experiment of sound localization/distance based on a WFS using a front and rear loudspeaker array system which is conducted on two listening points and shows average localization errors of 6.1° and 9.18°, while the average distance errors are –27% (0.5 m) and –29% (0.6 m), respectively.     

Thermoelectric Polymer Aerogels for Pressure–Temperature Sensing Applications

The evolution of the society is characterized by an increasing flow of information from things to the internet. Sensors have become the cornerstone of the internet‐of‐everything as they track various parameters in the society and send them to the cloud for analysis, forecast, or learning. With the many parameters to sense, sensors are becoming complex and difficult to manufacture. To reduce the complexity of manufacturing, one can instead create advanced functional materials that react to multiple stimuli. To this end, conducting polymer aerogels are promising materials as they combine elasticity and sensitivity to pressure and temperature. However, the challenge is to read independently pressure and temperature output signals without cross‐talk. Here, a strategy to fully decouple temperature and pressure reading in a dual‐parameter sensor based on thermoelectric polymer aerogels is demonstrated. It is found that aerogels made of poly(3,4‐ethylenedioxythiophene):poly(styrene sulfonate) (PEDOT:PSS) can display properties of semiconductors lying at the transition between insulator and semimetal upon exposure to high boiling point polar solvents, such as dimethylsulfoxide (DMSO). Importantly, because of the temperature‐independent charge transport observed for DMSO‐treated PEDOT‐based aerogel, a decoupled pressure and temperature sensing can be achieved without cross‐talk in the dual‐parameter sensor devices.     

非均匀二次声源分布下的2.5D高阶Ambisonics声场合成算法

为了提高感兴趣区域音源的声场合成质量,提出了一种非均匀二次声源分布的声场分析与合成方法.首先,将连续二次声源的采样分为两个不同的采样区域,在总扬声器数目一定的情况下,增大感兴趣区域的样点数目并减少感兴趣区域外样点数目.然后,利用采样混叠矩阵的Hermitian特性,使用瑞利熵来估计不同非均匀采样方案的最大误差,以指导非...     

扩声系统中设计、配置环绕和效果声应注意问题

说明了扩声系统设计中环绕声和效果声应注意的问题,指出了设计和配置环绕声（效果声）音箱的必要性,并介绍了如何设计和配置环绕声（效果声）音箱,为广大用户提供一个更加符合实际应用需要的扩声系统。     

听觉传输技术及其应用

文中综述了听觉传输技术的原理、应用和进展，在阐述了人类听觉系统对声源的定位机理后，讨论了实现听觉传输的基本原理和技术，包括双耳信号的检拾与模拟，双耳信号的扬声器重发，虚拟扬声器技术等；文中还探讨了听觉传输技术的局限性，综述了它在声学各方面的应用。文中最后还讨论了这方面今后的发展展望。     

A new configuration for echo canceller adaptable during double talkperiods

A three-port echo canceler (EC) configuration is proposed which observes the signal of the near-end side on a two-wire circuit in addition to the four-wire circuit signals. Embedding these signals on hybrid ports into a three-dimensional autoregressive process, echo path and innovations of near- and far-end speeches can be estimated through a three-channel lattice filter. The new configuration is then able to track echo path time variance, even during double talk (DT), and requires no changeover at either the beginning or end of DT, thus eliminating the need for DT detection. Two echo synthesizers utilizing inverse lattice and the echo path estimate possess guaranteed stability without the need for testing     

球源空间均匀分布阵的空间辐射场特性研究

定义了球源空间均匀分布阵的概念，从理论上推导了球源空间均匀分布阵远场声压和指向性公式，分析了它们与场点的位置、球源个数、相邻球源中心间距、波数及球源表面振速等因素的关系。并且对它们的关系作了仿真实验。在理论与仿真分析的基础上，给出了球源阵的设计原则：(1)根据球源空间阵大小的实际需要，确定球源个数和布阵方式；(2)球源本身的半径或体积尽可能选小一些，这样可减小阵的体积，以便尽可能增加球源个数并增强指向性；(3)在球源半径一定的条件下，相邻球源中心间距应尽量减小，最小到为球源直径的两倍。     

基于相位敏感光时域反射计的音频激励信号解调与复现

解调复现外部振动信号在实际工程监测应用中具有广阔的应用前景。基于相位敏感光时域反射计(Φ-OTDR)的DAS可以探测分布式单模光纤的振动信号,其集传感和光纤于一体且还原原始信号时无需参考信号,适用于工程应用。本文设计演示了一种基于 Φ-OTDR的直接监测的简易且低成本的音频振动信号的解调与复现系统。在实验装置中使用聚苯乙烯板作为声音在空气中传输的增敏材料。将正弦干扰声音、人声及音乐三种激励源通过扬声器对待测光纤播放,证明本系统能在10 km的传输距离中复现频率丰富的外部振动信号,且具有4 m的空间分辨率,该系统适用于长距离监测下的外部振动的解调复现。     

基于Windows平台的音效增强系统设计与实现

针对消费电子类产品中扬声器体积小型化所导致的低频重放能力有限、动态范围较小、频响曲线不佳、声场过窄等问题,开发了iSurround音效增强算法,包括扬声器均衡、图示均衡器、立体声增强、虚拟低音、动态范围压缩和自动音量控制等.在Windows平台上以虚拟声卡的形式实现该算法,最后运用主观听音实验的方法,证明了iSurround音效增强算法在低音表现力、声场宽度、空间环绕感等方面有明显提高.同时,也与目前主流音效算法就行了比较.     

利用参量阵扬声器进行有源噪声控制的研究

在有源噪声控制系统中,由于传统扬声器具有全指向性,在控制点处获得降噪效果的同时,在声场其他区域噪声会增大.参量阵扬声器是具有强指向性的新型声频声源,可以作为有源噪声控制系统中的次级声源,以实现特定方向或区域的降噪,同时减少对声场其他区域的影响.对参量阵扬声器和传统扬声器作为次级声源的有源噪声控制系统进行了仿真实验,结果...     

Cochannel interference reduction in optical PPM-CDMA systems

A multiple-user interference reduction technique is proposed for optical code-division multiple-access (CDMA) systems. Data symbols from each user are encoded using a pulse-position modulation (PPM) scheme before multiplexing. Modified prime sequences are adopted as the signature codes in the multiplexing process. An interesting property of this code is the uniformity of the cross correlation among its sequences. This property is the main key in constructing the multiple-access interference canceler. In addition to its simplicity, this canceler offers a great improvement in the error probability as compared to the system without cancellation. A simple modification to this canceler that enhances its performance is proposed as well     

声源空间分离对向前时域掩蔽阈值的影响

从实验上探讨了掩蔽和被掩蔽声源在水平面方向分离时的向前掩蔽效应,并与单扬声器重发方式作比较．得到了这2种情况下3种频率的被掩蔽纯音信号（250Hz,1170Hz,4000Hz）的掩蔽闻值随掩蔽窄带噪声5种声压级（30dB,40dB,50dB,60dB,70dB）变化的曲线。实验结果表明：声源空间分离时,除了个别情况外,掩蔽阈值都比单扬声器重发时有较大幅度的降低。文中进一步指出掩蔽阈值的降低是较优耳效应和双耳效应等的综合作用造成的。因此掩蔽声和被掩蔽声在空间上的分离会削弱向前时域掩蔽的作用,这一结论可为将来改进音频压缩算法提供理论基础。     

Some aspects of binaural sound

Experiments in binaural listening show that the two ears provide almost independent effects on the brain?namely, a sound message confined entirely to one ear does not interfere appreciably with a sound message confined entirely to the other ear. A result of this independence is that harmony, in a restricted sense, between binaurally heard tones does not exist. This fact might be used to create new effects in two-part music. In the most general sense, the experiments described in this article show that any two messages? whether in music, in words, or in symbols?can be separated more easily by the brain when the messages are heard binaurally, a finding that raises certain interesting implications.     

压缩式驱动器主观音质评价方法的比较

提出了一种基于感知模型的声辐射系统音质评价方法.采用此模型可应用Moore提出的参数以测试其是否适用于间接辐射扬声器的音质评价.对3种压缩式驱动器进行了测试,并且记录了它们在各种测试信号以及各种输入电压条件下的测试结果.墓于上述实验结果,从不同的角度提出了解决该问题的方法:(a)考虑经典的音质评价参数,如粗糙度、尖锐度...