有源头靠中一种分布式虚拟传声器优化方法

有源降噪头靠系统中,远程虚拟传声器技术能够解决控制点处与误差传声器处降噪量不匹配的问题。在实际应用中,多通道虚拟传声器技术存在收敛速度慢和运算复杂度高等问题。针对这个问题,本文通过重新设计远程虚拟传声技术的离线优化过程,提出一种分布式远程虚拟传声器技术优化方法。该方法将虚拟次级通路矩阵作对角化限制,同时对观测传递函数矩阵进行联合寻优,以实现一种分布式的更新算法。有源降噪头靠实验结果表明,所提算法能够有效降低远程虚拟传声器技术算法的运算复杂度,并且提升了算法的收敛速度。     

几种管道有源降噪器控制系统频率响应函数的确定

管道有源降噪器的控制系统根据传声器检测到的噪声信号,经过加工处理,产生激励次级声源的信号,使其产生的声波在下游与原来的噪声相抵消。控制系统应具备的频率响应函数取决于次级声源及检测传声器的配置,及其本身的频率响应。本文推导了在常用的几种不同次级声源和检测传声器配置下,所需的控制系统的频率响应函数;并介绍了一种测量传声器及扬声器在管道中的频率响应的方法。它们为控制系统的设计提供了依据。     

自适应零限波束形成语音增强算法鲁棒性分析

分析了双传声器自适应零限波束形成语音增强算法对传声器不一致和本底噪声的鲁棒性。结果表明:信干比越高,算法对传声器不一致的鲁棒性越差;当信干比很低时,算法对相位不一致是鲁棒的。幅度不一致会降低算法对干扰的抑制能力,但引起的目标信号失真很小。相对于传声器不一致,本底噪声对算法性能的影响较小。对自适应滤波器权值加约束和对传声器做校准可以提高算法性能。     

虚拟声屏障在变压器低频降噪中的实验研究

针对开口房间内的变压器,在开口处布放若干扬声器和误差传声器构成虚拟声屏障,实验研究了虚拟声屏障对通过开口向外辐射的低频线谱噪声的控制效果。将15个次级源近似均匀分布在面积为2 m′ 2.7 m的开口面上,左右间距约58.5 cm、上下间距45-65 cm,15个误差传声器分别位于对应的次级源正前方1 m,系统采用自适应谐波降噪算法。结果表明：虚拟声屏障系统在误差点100 Hz、200 Hz和300 Hz的平均降噪量分别达到12.7 dB、19.9 dB和22.2 dB,虚拟声屏障对100 Hz、200 Hz、300 Hz线谱噪声的控制效果与单层封闭窗户相当,且内部合成参考信号,无需外接参考传声器。采用虚拟声屏障对开口房间内的变压器降噪的好处是实现室内外的自然通风,便于变压器的散热。     

一种基于虚拟传感的无需误差传声器的自适应有源降噪方法*

该文基于虚拟传感技术引入了一种用于耳机的无需误差传声器的自适应有源降噪方法。该算法仅使用一个参考传声器实现了一种前馈和反馈自适应算法结合的有源降噪算法,提高了有源降噪稳定性,简化了耳机硬件结构。利用DSP平台实现了该文提出的方案,并通过实验验证了其良好的降噪性能和实用价值。     

一种有源头靠系统中虚拟传声器技术的改进方法

对人头移动导致有源头靠降噪效果下降的问题进行了理论分析,得到了残余噪声的预测表达式,提出了一种通过设计次级通道模型改进虚拟传声器技术性能的方法。分析表明,人耳位置处降噪量取决于虚拟传声器方法中采用的次级通道模型与人头移动到不同位置处的真实次级通道之间的匹配程度。通过设计合理的次级通道模型,可有效扩大人耳获得有效降噪量的人头允许移动范围。仿真分析和实验结果验证了该方法的有效性。     

IEPnP:一种基于EPnP的相机位姿迭代估计算法

近年来,EPnP算法作为一种相机位姿估计的解析算法,因其较低的计算复杂度而得到广泛的关注,但该算法对图像噪声的稳健性不强。提出了一种基于EPnP算法的迭代算法,即IEPnP算法。IEPnP算法保留了EPnP算法的主要思想,构造了4个虚拟控制点,利用弱透视投影模型获得相机的初始位姿,计算出虚拟控制点在相机坐标系下的坐标,然后通过高斯-牛顿法对虚拟控制点在相机坐标系下的坐标进行优化求解,最终通过解决绝对定向问题来获得对相机位姿的估计。IEPnP算法简化了EPnP算法的计算过程。在不同的图像噪声水平下进行仿真实验,结果表明,相比于EPnP算法,IEPnP算法不仅保持了较高的计算效率,而且对图像噪声具有更强的稳健性。     

球谐域类正则化宽带超指向性波束形成算法

提出一种球谐域类正则化宽带超指向性波束形成算法,通过结合超指向性波束形成器与延迟求和波束形成器控制阵列白噪声增益和指向性因数,推导类正则化宽带超指向性波束形器的球谐域表达式。在此基础上设计新的可控指向性因数波束形成器,可在不显著放大低频白噪声的前提下实现设定的波束宽度。采用32元球阵的仿真结果表明,对于语音测试场景,类正则化波束形成器的最高PESQ得分相比于传统的超指向性波束形成器与延迟求和波束形成器分别提高了约0.5和0.4,平均词错率分别减少了约9.5%和8.1%。主观测试实验也表明类正则化波束形成器在方向性噪声和扩散场噪声环境下都可以获得更好的主观听觉感受。利用实验数据对算法性能进行测试,实验结果同样验证了该方法在实际声学环境中的有效性。      

使用脉冲声和矢量传声器的现场测量吸声系数

使用不同声源利用矢量传声器对毛毡材料进行现场吸声系数测试,研究了不同背景下不同声源的抗噪能力。矢量传声器可以同时测得声压和质点振速信号,进而可计算得到阻抗,利用自由空间和材料表面的阻抗可计算得到材料的吸声系数。在此次实验中,使用不同声源分别在无干扰和有一白噪声干扰源的两种情况下进行测试。结果表明,使用对数扫频脉冲和巴特沃斯脉冲测试所得的吸声系数曲线更平滑,说明脉冲声可以有效降低环境反射的影响,在高噪声背景下使用对数扫频脉冲测试所得的结果基本没有受到背景噪声的影响,说明对数扫频脉冲的抗噪能力更强。因此,使用对数扫频脉冲作为声源进行测试可有效减弱环境反射和背景噪声的影响。     

声矢量锥形阵的高阶累积量波达方向估计

为解决信源在较低信噪比情况下的测向分辨率问题,提出阵列可扩展的声矢量锥形阵测向算法。算法基于四阶累积量的阵列扩展和高斯噪声抑制特性,计算声矢量传声器不同输出分量的四阶累积量,使其在三维方向上扩展与原阵型结构相同的虚拟阵,从而构造包含角度信息的旋转不变矩阵进行测向。推导给出了算法的克拉美罗界,理论分析了算法性能受信噪比、采样快拍以及入射声源俯仰角的影响。仿真实验验证了该算法较常规声矢量阵ESPRIT算法有更优的噪声抑制能力及更高分辨的DOA估计性能。      

基于双传声器的蓝牙耳机降噪算法

用于免提通信设备的语音增强算法一直是研究的热点问题,而算法处理结果的音质问题近年来也备受关注。针对基于双传声器降噪的蓝牙耳机系统,将常用多通道传声器降噪算法归纳为基于相干函数法和基于空间预分离法这两大类进行分析和比较。基于相干函数法利用两个通道间信号的相干函数对含噪信号滤波达到降噪目的,而基于空间预分离法利用空间特性从含噪信号中分离出噪声参考信号来消除噪声。分析基于降噪量、语音音质和综合性能三个指标,从约束语音损伤的角度分析最优解的形式,并对比两类算法的实际性能。结果表明选择合适的算法可权衡降噪量与语音损伤,达到较好的综合性能。     

A nonlinear active noise control algorithm for virtual microphones controlling chaotic noise

In active noise control (ANC) systems, virtual microphones provide a means of projecting the zone of quiet away from the physical microphone to a remote location. To date, linear ANC algorithms, such as the filtered-x least mean square (FXLMS) algorithm, have been used with virtual sensing techniques. In this paper, a nonlinear ANC algorithm is developed for a virtual microphone by integrating the remote microphone technique with the filtered-s least mean square (FSLMS) algorithm. The proposed algorithm is evaluated experimentally in the cancellation of chaotic noise in a one-dimensional duct. The secondary paths evaluated experimentally exhibit non-minimum phase response and hence poor performance is obtained with the conventional FXLMS algorithm compared to the proposed FSLMS based algorithm. This is because the latter is capable of predicting the chaotic signal found in many physical processes responsible for noise. In addition, the proposed algorithm is shown to outperform the FXLMS based remote microphone technique under the causality constraint (when the propagation delay of the secondary path is greater than the primary path). A number of experimental results are presented in this paper to compare the performance of the FSLMS algorithm based virtual ANC algorithm with the FXLMS based virtual ANC algorithm.     

Effects of directional microphone and adaptive multichannel noise reduction algorithm on cochlear implant performance

Although cochlear implant (CI) users have enjoyed good speech recognition in quiet, they still have difficulties understanding speech in noise. We conducted three experiments to determine whether a directional microphone and an adaptive multichannel noise reduction algorithm could enhance CI performance in noise and whether Speech Transmission Index (STI) can be used to predict CI performance in various acoustic and signal processing conditions. In Experiment I, CI users listened to speech in noise processed by 4 hearing aid settings: omni-directional microphone, omni-directional microphone plus noise reduction, directional microphone, and directional microphone plus noise reduction. The directional microphone significantly improved speech recognition in noise. Both directional microphone and noise reduction algorithm improved overall preference. In Experiment II, normal hearing individuals listened to the recorded speech produced by 4- or 8-channel CI simulations. The 8-channel simulation yielded similar speech recognition results as in Experiment I, whereas the 4-channel simulation produced no significant difference among the 4 settings. In Experiment III, we examined the relationship between STIs and speech recognition. The results suggested that STI could predict actual and simulated CI speech intelligibility with acoustic degradation and the directional microphone, but not the noise reduction algorithm. Implications for intelligibility enhancement are discussed.     

Experimental evaluation of leaky least-mean-square algorithms for active noise reduction in communication headsets

An adaptive leaky normalized least-mean-square (NLMS) algorithm has been developed to optimize stability and performance of active noise cancellation systems. The research addresses LMS filter performance issues related to insufficient excitation, nonstationary noise fields, and time-varying signal-to-noise ratio. The adaptive leaky NLMS algorithm is based on a Lyapunov tuning approach in which three candidate algorithms, each of which is a function of the instantaneous measured reference input, measurement noise variance, and filter length, are shown to provide varying degrees of tradeoff between stability and noise reduction performance. Each algorithm is evaluated experimentally for reduction of low frequency noise in communication headsets, and stability and noise reduction performance are compared with that of traditional NLMS and fixed-leakage NLMS algorithms. Acoustic measurements are made in a specially designed acoustic test cell which is based on the original work of Ryan et al. ["Enclosure for low frequency assessment of active noise reducing circumaural headsets and hearing protection," Can. Acoust. 21, 19-20 (1993)] and which provides a highly controlled and uniform acoustic environment. The stability and performance of the active noise reduction system, including a prototype communication headset, are investigated for a variety of noise sources ranging from stationary tonal noise to highly nonstationary measured F-16 aircraft noise over a 20 dB dynamic range. Results demonstrate significant improvements in stability of Lyapunov-tuned LMS algorithms over traditional leaky or nonleaky normalized algorithms, while providing noise reduction performance equivalent to that of the NLMS algorithm for idealized noise fields.     

调节水床效应的双梯度有源噪声控制自适应算法*

为了使自适应反馈有源噪声控制系统能够兼顾水床效应,通过约束次级信号的能量以调节噪声放大,设计了一种双梯度算法。当次级信号满足约束,算法沿着最小化误差信号能量的梯度方向迭代,反之,则沿着最小化次级信号能量的方向迭代。在有源降噪耳机实例中的对比结果表明,该算法能调节噪声放大并保持较大的降噪带宽,且运算量没有显著增加。     

Evaluation of a noise reduction system for the assessment of click-evoked otoacoustic emissions

A recently proposed noise reduction system intended to facilitate the assessment of click-evoked otoacoustic emission (CEOAE) in noisy environments [Comput. Biol. Med. 30, 341 (2000)] is evaluated using 13 normally hearing ears and 9 ears with a sensorineural hearing loss. The noise reduction system is based on an adaptive noise canceller design using an additional noise-only reference microphone and intended to reduce externally generated noise. The system is tested in quiet and at different levels of white noise. The three main design parameters of the noise reduction system (adaptation time constant, length of the adaptive filter, and position of the noise reference microphone) are varied systematically in different experiments. With the noise reduction system active, CEOAE can be assessed correctly at noise levels which are 5 to 9 dB higher than without the noise reduction system. For the range of adaptation time constants considered (65.6 to 656 ms), no statistically significant effect on the amount of noise reduction is observed. Noise reduction is highest when the reference microphone is positioned close to the ear probe. Using this reference microphone position and adaptive filters of 6.56 ms in length, average noise reductions of 7.17 to 8.50 dB are achieved.     

Causality study on a feedforward active noise control headset with different noise coming directions in free field

A systematic analysis is proposed to predict the performance of a typical feedforward single channel ANC headset in terms of the delay, especially the non-causal delay caused by different noise coming directions. First, the performance of a non-causal feedforward system for a band-limited noise is analyzed by using a simplified pure delay model, where it is found that the noise reduction bandwidth is narrowed and the maximum noise reduction is decreased with the increase of the non-causal delay. Second, a systematic method is developed, which can be used to predict the system performance with measured primary and secondary path transfer functions in most practical sound fields and to study the effects of the control filter length and the path delay on the performance. Then, the causality of a typical feedforward active noise control headset with the primary source at 0° and 90° positions in an anechoic chamber is analyzed, and the performance for the two locations predicted by the systematic analysis is shown in good agreements with the experiment results. Finally, an experiment of a typical feedforward active noise control headset in a reverberation chamber is carried out, which shows the validity of the proposed systematic analysis for other more practical sound fields.     

Novel active noise-reducing headset using earshell vibration control

Active noise-reducing (ANR) headsets are available commercially in applications varying from aviation communication to consumer audio. Current ANR systems use passive attenuation at high frequencies and loudspeaker-based active noise control at low frequencies to achieve broadband noise reduction. This paper presents a novel ANR headset in which the external noise transmitted to the user's ear via earshell vibration is reduced by controlling the vibration of the earshell using force actuators acting against an inertial mass or the earshell headband. Model-based theoretical analysis using velocity feedback control showed that current piezoelectric actuators provide sufficient force but require lower stiffness for improved low-frequency performance. Control simulations based on experimental data from a laboratory headset showed that good performance can potentially be achieved in practice by a robust feedback controller, while a single-frequency real-time control experiment verified that noise reduction can be achieved using earshell vibration control.