基于脉冲声分离的刚性球散射声控制

提出了一种基于脉冲声的三维空间中刚性球散射声分离方法,并利用前馈、固定系数控制方式对分离出的散射声进行有源控制,抑制散射声强度,实现了刚性球散射体在观测点处“声学不可见”。该方法利用脉冲信号作为初级噪声,通过有无刚性球时传声器采集脉冲信号的差值确定散射声大小,实现散射声与声源直达声的分离。对分离出的散射声进行多通道有源控制以验证该文所提分离方法及控制系统的有效性。实验结果表明,700~1000 Hz范围内,有源控制开启后,双通道散射声的平均降噪量大于5 dB,多通道散射声的平均降噪量大于8 dB,且误差传声器处采集的残余声场与无刚性球时采集的初级声场信号波形基本一致,实现了刚性球散射体在误差传声器处“声学不可见”。此外,参考传声器布放位置的选取问题也在该文做了详细讨论。     

基于双传声器对的多声源二维定位跟踪算法

提出一种房间混响声场环境下的多声源二维定位跟踪算法。研究了基于盲源分离的时延估计,以及联合空间分布的多个传声器对的定位算法。用高斯似然函数解决在多源、多维情况下声源定位的时延匹对模糊问题,使之能够用双传声器对实现对多个声源的二维定位,结合粒子滤波算法实现对多个运动声源的跟踪。仿真实验验证了提出算法的有效性。      

基于自适应滤波器的大气湍流噪声抑制

为了解决大气湍流效应所引起的光无线通信系统中的衰落问题,建立了与大气湍流噪声及其它加性噪声有关的信号模型,实现了最大输出信噪比可提高3-20dB的匹配滤波器。根据最大似然比准则,推导了强度调制/直接探测(IM/DD)的光无线通信系统的最佳判决门限。采用自适应滤波器实现了对探测门限中所含的信号和湍流的统计值的预测,该自适应门限探测技术对大气湍流噪声有很好的抑制作用,在强湍流情况下,可降低系统的误码率到10-5以下。     

基于最差性能优化的运动声源稳健聚焦定位识别方法研究

基于被动合成孔径原理,提出了一种具有高稳健性的运动声源高分辨聚焦定位识别方法.该方法采用综合优化手段,通过矢量最大似然聚焦定位算法生成虚拟阵列坐标及数据矩阵,进而利用基于最差性能优化的稀疏虚拟阵列聚焦算法,获取稳健的高分辨定位识别效果.理论及仿真研究表明,该方法对于非匀速运动以及与基阵存在运动倾角的复杂情况具有较强的适用性,聚焦空间谱表现出更大的动态范围、更为尖锐的聚焦峰尺度以及更强的背景噪声起伏压制能力.湖上试验进一步验证,在高分辨最小方差信号无畸变响应法(MVDR)聚焦算法动态范围仅为3.5 dB的相 关键词： 稳健性 运动噪声源定位识别 矢量阵 最差性能优化     

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.     

A fast signal subspace approach for the determination of absolute levels from phased microphone array measurements

Phased microphone arrays are used in a variety of applications for the estimation of acoustic source location and spectra. The popular conventional delay-and-sum beamforming methods used with such arrays suffer from inaccurate estimations of absolute source levels and in some cases also from low resolution. Deconvolution approaches such as DAMAS have better performance, but require high computational effort. A fast beamforming method is proposed that can be used in conjunction with a phased microphone array in applications with focus on the correct quantitative estimation of acoustic source spectra. This method bases on an eigenvalue decomposition of the cross spectral matrix of microphone signals and uses the eigenvalues from the signal subspace to estimate absolute source levels. The theoretical basis of the method is discussed together with an assessment of the quality of the estimation. Experimental tests using a loudspeaker setup and an airfoil trailing edge noise setup in an aeroacoustic wind tunnel show that the proposed method is robust and leads to reliable quantitative results.     

A comparison of filter design structures for multi-channel acoustic communication systems

The application of inverse filter designs as a means of providing improved communication performance in acoustic environments is investigated. Tikhonov regularized inverse filters of channel transfer functions calculated in the frequency domain are used as a means of obtaining multi-channel filters. Three classifications of inverse filter structures have been considered using time-domain simulations. The performance of Tikhonov regularized inverse filters designed according to each of these classifications is compared with each other and against a filter design developed by Stojanovic [Stojanovic, M. (2005). "Retrofocusing techniques for high rate acoustic communications," J. Acoust. Soc. Am. 117, 1173-1185]. It is shown that the filter design developed by Stojanovic requires less regularization and outperforms the Tikhonov regularized inverse filter designs when communicating over a single channel. While the filter developed by Stojanovic is designed to use multiple transmitters to transmit to a single receiver, the filter was implemented in a multi-channel system and proposed to have a focusing similar to that obtained using time-reversal. It was found that for the scenario used in the simulation, the Tikhonov regularized inverse design for full multi-channel inversion achieved better focusing than the design by Stojanovic, where simulation results show 20 dB less cross-talk at the expense of around 2 dB loss in signal strength.     

基于音视频信息融合的目标检测与跟踪算法

针对单一视觉跟踪算法易受遮挡影响的缺陷,提出一种基于音视频信息融合的目标检测与跟踪算法。整个算法框架包括视频检测与跟踪、声源定位、音视频信息融合跟踪3个模块。视频检测与跟踪模块采用YOLOv5m算法作为视觉检测的框架,使用无迹卡尔曼滤波和匈牙利算法实现多目标的跟踪与匹配;声源定位模块采用十字型麦克风阵列获取音频信息,结合各麦克风接收信号的时延计算声源方位;音视频信息融合跟踪模块构建音视频似然函数和音视频重要性采样函数,采用重要性粒子滤波作为音视频融合跟踪的算法,实现对目标的跟踪。在室内复杂环境下对算法性能进行测试,结果表明该算法跟踪准确率达到90.68%,相较于单一模态算法具有更好的性能。     

Beamforming correction for dipole measurement using two-dimensional microphone arrays

In this paper, a beamforming correction for identifying dipole sources by means of phased microphone array measurements is presented and implemented numerically and experimentally. Conventional beamforming techniques, which are developed for monopole sources, can lead to significant errors when applied to reconstruct dipole sources. A previous correction technique to microphone signals is extended to account for both source location and source power for two-dimensional microphone arrays. The new dipole-beamforming algorithm is developed by modifying the basic source definition used for beamforming. This technique improves the previous signal correction method and yields a beamformer applicable to sources which are suspected to be dipole in nature. Numerical simulations are performed, which validate the capability of this beamformer to recover ideal dipole sources. The beamforming correction is applied to the identification of realistic aeolian-tone dipoles and shows an improvement of array performance on estimating dipole source powers.     

用于自适应波束形成语音增强的球谐域掩蔽函数估计方法

提出一种用于球形阵列自适应波束形成的掩蔽函数估计方法.该方法利用包含空间信息的球谐系数提取低维空间向量,并采用复高斯混合模型和深度学习两种方案来估计掩蔽函数,最终利用估计的掩蔽函数设计最小方差无失真响应波束形成器,以达到空域滤波的效果.理论分析和仿真实验证明,对于相同时长的声信号,球谐域掩蔽函数估计方法的计算复杂度比传...     

Reconstruction of the signal produced by a directional sound source from remote multi-microphone recordings

A mathematical method for reconstructing the signal produced by a directional sound source from knowledge of the same signal in the far field, i.e., microphone recordings, is developed. The key idea is to compute inverse filters that compensate for the directional filtering of the signal by the sound source directivity, using a least-square error optimization strategy. Previous work pointed out how the method strongly depends on arrival times of signal in the microphone recordings. Two strategies are used in this paper for calculating the time shifts that are afterward taken as inputs, together with source directivity, for the reconstruction. The method has been tested in a laboratory environment, where ground truth was available, with a Polaroid transducer as source. The reconstructions are similar with both strategies. The performance of the method also depends on source orientation.     

Time-delay estimation of reverberated speech exploiting harmonic structure.

The relative time delay associated with a speech signal received at a pair of spatially separated microphones is a key component in talker localization and microphone array beamforming procedures. The traditional method for estimating this parameter utilizes the generalized cross correlation (GCC), the performance of which is compromised by the presence of room reverberations and background noise. Typically, the GCC filtering criteria used are either focused on the signal degradations due to additive noise or those due exclusively to multipath channel effects. There has been relatively little success at applying GCC weighting schemes which are robust to both of these conditions. This paper details an alternative approach which attempts to employ a signal-dependent criterion, namely, the estimated periodicity of the speech signal, to design a GCC filter appropriate for the combination of noise and multipath distortions. Simulations are performed across a range of room conditions to illustrate the utility of the proposed time-delay estimation method relative to conventional GCC filtering approaches.     

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.     

A directional acoustic array using silicon micromachined piezoresistive microphones

The need for noise source localization and characterization has driven the development of advanced sound field measurement techniques using microphone arrays. Unfortunately, the cost and complexity of these systems currently limit their widespread use. Directional acoustic arrays are commonly used in wind tunnel studies of aeroacoustic sources and may consist of hundreds of condenser microphones. A microelectromechanical system (MEMS)-based directional acoustic array system is presented to demonstrate key technologies to reduce the cost, increase the mobility, and improve the data processing efficiency versus conventional systems. The system uses 16 hybrid-packaged MEMS silicon piezoresistive microphones that are mounted to a printed circuit board. In addition, a high-speed signal processing system was employed to generate the array response in near real time. Dynamic calibrations of the microphone sensor modules indicate an average sensitivity of 831 microV/Pa with matched magnitude (+/-0.6 dB) and phase (+/-1 degree) responses between devices. The array system was characterized in an anechoic chamber using a monopole source as a function of frequency, sound pressure level, and source location. The performance of the MEMS-based array is comparable to conventional array systems and also benefits from significant cost savings.     

Experimental determination of cascade parameters of a hearing-aid microphone via the two-load method

Presented in this article is a computer-aided experimental method for obtaining the cascade parameters of the two-port model of a miniature hearing-aid microphone. The method is an adaptation of the "two-load" method [D.P. Egolf and R.G. Leonard, J. Acoust. Soc. Am. 62, 1013-1023 (1977)] to acoustoelectric, rather than electroacoustic, transducers. The cascade parameters of a particular microphone, determined by this method, were within 2.5 dB of the manufacturer's published open-circuit sensitivity data. In an attempt to further verify the numerical cascade-parameter data, a two-port model of the microphone was used to simulate experimental voltages developed across two different complex electrical load impedances attached to the microphone. The results showed experimental/simulation differences of no greater than 3.0 dB at any frequency. The two-port microphone model and associated cascade parameters are currently being incorporated into a computer-based plan for mathematical simulation of an entire in situ hearing aid.     

A selective two microphone acoustic intensity method

A multiple input, two output model is proposed which enables the two microphone acoustic intensity method to decompose the intensity vector into contributions from individual sources, even when they are coupled and in close proximity within the measurement surface. By treating characteristic signals from each source as the inputs, and the sound pressure signals from the two closely spaced microphones as the outputs, the model's frequency response functions are developed from a least squares approximation. The cross spectrum between the two microphones can then be expressed as a function of the input signal spectra and the model's frequency response functions. By manipulating the model terms the selective cross spectrum associated with the radiation from each individual source can then be estimated. The selective cross spectrum is then processed via standard methods to obtain the acoustic intensity vector from each source. A series of laboratory experiments is summarized which demonstrates that the technique can accurately decompose the acoustic intensity vector from highly coherent sources (γ₁₂² > 0·9) buried in background noise in a semireverberant environment, to within 1 dB of the directly measured intensities.     

基于掺铒光纤/喇曼混合放大的光纤激光器布喇格光栅传感系统

提出了一种提高长距离光纤布喇格光栅信噪比以进行准分布测量的新方法.该方法基于掺铒光纤/喇曼混合放大的光纤激光器结构,掺铒光纤和滤波器构成的环形结构产生激光作为光源,喇曼光纤放大器对布喇格光栅信号进行低噪音的双向放大,置于远处的掺铒光纤利用剩余的泵浦功率产生自发辐射光和放大传感信号,为远处掺铒光纤之后的布喇格光栅传感器提供信号光以及补偿由于长距离传输造成的光纤损耗.实验显示,与使用宽带光源的传感方式相比,系统的性能得到显著提高,仅使用小功率泵浦,分布在50 km光纤上的FBG均获得了超过58 dB的优良信噪比.     

Calibration of the pressure sensitivity of microphones by a free-field method at frequencies up to 80 khz

A free-field (FF) substitution method for calibrating the pressure sensitivity of microphones at frequencies up to 80 kHz is demonstrated with both grazing and normal-incidence geometries. The substitution-based method, as opposed to a simultaneous method, avoids problems associated with the nonuniformity of the sound field and, as applied here, uses a 1/4-in. air-condenser pressure microphone as a known reference. Best results were obtained with a centrifugal fan, which is used as a random, broadband sound source. A broadband source minimizes reflection-related interferences that can plague FF measurements. Calibrations were performed on 1/4-in. FF air-condenser, electret, and microelectromechanical systems (MEMS) microphones in an anechoic chamber. The uncertainty of this FF method is estimated by comparing the pressure sensitivity of an air-condenser FF microphone, as derived from the FF measurement, with that of an electrostatic actuator calibration. The root-mean-square difference is found to be +/- 0.3 dB over the range 1-80 kHz, and the combined standard uncertainty of the FF method, including other significant contributions, is +/- 0.41 dB.     

传声器阵列对高频弱声源的声成像识别方法

研究了传声器阵列对高频弱声源的识别定位方法。该方法根据高频声源的指向性和阵列探测特性等特点,提出了利用信噪比加权方法提高有效阵元对声成像的贡献,根据信噪比的大小对每个阵元添加不同的权值,可以显著提高传声器阵列对高频弱声源的声像清晰度。仿真分析了阵元加权和不加权两种方法对阵列声成像结果的影响,以某型号笔记本电脑电路板噪声为对象进行的实验表明,在阵列测量中充分利用有效阵元信号可以实现对声压级低达10~20dB的微弱噪声源的精确测量。      

Numerical simulation of transfer and attenuation characteristics of soft-tissue conducted sound originating from vocal tract

A non-audible murmur (NAM), a very weak speech sound produced without vocal cord vibration, can be detected by a special NAM microphone attached to the neck, thereby providing a new speech communication tool for functional speech disorders as well as human-to-machine and human-to-human interfaces with inaudible voice input for use with unimpaired. The NAM microphone is a condenser microphone covered with soft-silicone impression material that provides good impedance matching with the soft tissues of the neck. Because higher-frequency components are suppressed severely, however, the NAM detected with this device can be insufficiently clear. To improve NAM clarity, the mechanism of NAM production as well as the transfer characteristics of the NAM in soft neck tissues must be clarified. We have investigated sound propagation from the vocal tract to the neck surface, using a finite difference time domain method and a head model based on magnetic resonance imaging scans. Numerical results show that, compared to air-conducted sound detected in front of a mouth, soft-tissue-conducted sound attenuates 50 dB at 1 kHz, which consists of 30 dB full-range attenuation due to air-to-soft-tissues transmission loss and −10 dB/octave spectral decay due to a propagation loss in soft tissues. The decay agrees well with the spectral characteristics of the measured NAM.