Spatial filtering of sound beams by sonic crystals

We propose and numerically demonstrate the feasibility of spatial filtering, i.e. a selective removal of the high spatial frequency components, of narrow acoustic beams during their propagation along a two-dimensional sonic crystal. The filtering effect occurs due to two different mechanisms, depending on the particular form of the dispersion surfaces of the propagation modes in sonic crystal. Reflection or deflection of undesired spatial frequency components occurs for these two mechanisms. It is shown that, as a consequence of spatial filtering, the angular spectrum of the beams is narrowed, and the noisy beams are efficiently cleaned.     

Active sound control by directional sources in the free field

TheprojectissupportedbyAeronauticalScienceFoundation.I.IntroductionThecontro1ofsoundfieldbyintroducingacousticsourcesisatopicofnoisecontrolinwhichanincrcasinginterestisbeingshown.Gcnerallytherearemain1ytwoaspectsinthere-searchesonActiveNoiseContro1(ANC)--theoriesofactivesoundattenuationwhichisthebasementofthetCchnique,anddesignofthecontrolsystem.Recent1y,anewtheorywhichisconsideredtohewc11suitcdtotheanalysisofpracticalproblemshasbeenpresentedbyP.A.Nc1sonetal.[ll.Itshowsthattheminimumpowe…     

仿人耳听觉的助听器双耳声源定位算法

为提高复杂场景下的听障患者的语言理解度,本文提出一种仿人耳听觉的助听器双耳声源定位算法。算法首先借鉴耳蜗分频特性和听觉掩蔽特性,将声音信号进行多通道分解,并提取人耳敏感频带的信号进行双耳时间差(Interaural Time Difference,ITD)估计;然后基于人耳哈斯效应,提取有效的ITD信息;最后采用头相关模型,将ITD转化为声源方向信息。同时,为了改善混响和多干扰声场景下的声源定位能力,本文提出一种多通道的加权联合策略。仿真和场景测试实验表明,算法的抗干扰性强,定位精度高。而且,在7名受试者的理解度测试中,同现有的助听器增强算法相比,结合定位算法的语音增强算法达到3~5dB的性能改善。     

仿人耳听觉的助听器双耳声源定位算法

为提高复杂场景下的听障患者的语言理解度,本文提出一种仿人耳听觉的助听器双耳声源定位算法。算法首先借鉴耳蜗分频特性和听觉掩蔽特性,将声音信号进行多通道分解,并提取人耳敏感频带的信号进行双耳时间差(Interaural Time Difference,ITD)估计;然后基于人耳哈斯效应,提取有效的ITD信息;最后采用头相关模型,将ITD转化为声源方向信息。同时,为了改善混响和多干扰声场景下的声源定位能力,本文提出一种多通道的加权联合策略。仿真和场景测试实验表明,算法的抗干扰性强,定位精度高。而且,在7名受试者的理解度测试中,同现有的助听器增强算法相比,结合定位算法的语音增强算法达到3~5 dB的性能改善。     

基于声传播算子的声源定位实验模拟方法研究

声传播算子是一种高效的时域声场计算方法,它能够很方便地计算出给定系统参数下任意时刻任意位置的声场变化情况,本文采用这种方法计算所得的二维房间声场信息进行传声器阵列的声源定位仿真实验。计算结果表明,用该方法获取的阵列数据能有效地应用于阵列信号处理算法中,准确地估计出初始高斯脉冲声源的方向。声传播算子声场计算方法能为传声器阵列声源定位的实验提供方便,使得传声器阵列声源定位算法在不同混响时间的鲁棒性实验研究变得更加简捷。     

On the accuracy of the ray-tracing algorithms based on various sound receiver models

In this paper, the effect of sound ray detection on the accuracy of sound field simulation by using the ray-tracing technique has been discussed. The current methods to define the receiver size have been briefly described. Then a new equation for the computation of the size of a receiver has been proposed and it has considered such factors as the size of the room, the number of initial rays and the distance from source to receiver. The corresponding algorithm has been compared with two other algorithms that are based on different receiver models. It can be concluded from the computed and measured results that the new model is applicable for more extensive situations.     

运动声源快速定位的声达时差法

针对声达时差法只能用于非运动声源定位的问题,本文提出一种运动声源快速定位方法。该方法以声达时差为基本定位原理,基于声源计算位置对多普勒效应进行解耦并进行声信号多普勒效应修正,根据三角定位方法构建声传播空间矩阵,以声源位置偏差度为目标基于单纯形优化搜索算法进行声源位置快速逼近,实现了对匀速直线运动的单声源的定位追踪,提高定位实时性。该方法将声达时差法拓展到运动声源的定位,同时解决了消除多普勒效应带来的计算过程复杂、运算量大的问题,仅用4个传声器就可实现运动声源的快速定位,突破了传统运动声源识别中对大传声器阵列的依赖。仿真实验和实车运动声源识别实验结果证明了该方法的有效性,本研究为短时发声运动声源的识别提供了一种简便、高效的方法。     

基于增强GA-BP神经网络的软件错误定位方法

在软件开发和后期维护的过程中,进行软件调试来定位错误并修正错误是其中最复杂且成本最大的一部分;文章针对现有基于神经网络的软件错误定位方法中的权值和阈值设定不方便、鲁棒性差等问题,结合正交实验设计思想和遗传算法(Genetic Algorithm),提出了一种基于增强遗传BP神经网络的软件错误定位方法;并将其同基于GA-BP神经网络的和基于BP神经网络的定位方法都在MATLAB上进行了实验,实验数据来源西门子测试集,从结果上看,基于增强GA-BP神经网络的软件错误定位方法在定位错误的效率和精确度上都有一些进步。     

Estimation of interaural level difference based on anthropometry and its effect on sound localization

Individualization of head-related transfer functions (HRTFs) is important for highly accurate sound localization systems such as virtual auditory displays. A method to estimate interaural level differences (ILDs) from a listener's anthropometry is presented in this paper to avoid the burden of direct measurement of HRTFs. The main result presented in this paper is that localization is improved with nonindividualized HRTF if ILD is fitted to the listener. First, the relationship between ILDs and the anthropometric parameters was analyzed using multiple regression analysis. The azimuthal variation of the ILDs in each 1/3-octave band was then estimated from the listener's anthropometric parameters. A psychoacoustical experiment was carried out to evaluate its effectiveness. The experimental results show that the adjustment of the frequency characteristics of ILDs for a listener with the proposed method is effective for localization accuracy.     

基于桥接Transformer的小样本优化鸟声识别网络

针对实际鸟类监测环境中,收集鸟声声频数据分布不均匀,导致神经网络训练不充分,分类识别测试准确率低的问题,设计了一种桥接Transformer神经网络模型。该网络首先利用原始鸟声声频信号生成短时傅里叶变换语谱图作为输入特征,之后将语谱图输入到由注意力模块和卷积模块桥接组成的Transformer网络中,完成对语谱图中全局特征和局部特征的信息交互,最后利用单层Transformer编码器实现对每一个批次样本的损失优化,得到最终的分类结果。在Birdsdata和xeno-canto鸟声数据集上进行小样本实验,分别获得了91.34%和82.63%的平均准确率,与其他鸟声识别网络进行了对比实验,验证了该网络的有效性。     

圆柱绕流气动声的偶极子及四极子源法定量研究

以亚临界三维圆柱绕流的气动噪声为对象,研究声类比理论中偶极子及四极子源模型在预测低Mach数流动气动声的可靠性及准确性。使用大涡模拟(LES)得到非定常流场,并依据声类比中的Curle等效偶极子面源和Lighthill四极子体源模型,提取相应的声源数据,经Fourier变换得到涡脱落频率处的声源信息,进而定量预测圆柱绕流的气动声。结果表明:Curle模型的结果与实验结果吻合良好,Lighthill体源模型预测的准确性依赖于声源区域截断,不恰当的声源截断将导致错误的声场预测。     

Improved steered response power method for sound source localization based on principal eigenvector

Sound source localization is essential in many microphone arrays application, ranging from teleconferencing systems to artificial perception in a reverberant noisy environment. The steered response power (SRP) using the phase transform (SRP-PHAT) source localization algorithm has been proved robust, however, the performance of the SRP-PHAT algorithm degrades in highly reverberant noisy environment. Though the SRP-based maximum likelihood localizers are more robust than SRP-PHAT, they have the drawback of requiring noise variance to be estimated in a silent room. This paper presents an improved SRP-PHAT algorithm based on principal eigenvector. Sound source location is estimated from the principal eigenvector computed from the frequency-domain correlation matrix. Using both simulated and real data, we show that the proposed algorithm achieves higher source localization accuracy compared to the SRP-PHAT algorithm.     

基于Group Lasso的多重信号分类声源 定位优化算法*

多重信号分类算法因其抑制噪声能力强、计算速度快等优点,在声源定位领域得到广泛应用。但该算法在中低频段分辨率及聚焦性能较差。针对该问题,提出一种基于Group Lasso的多重信号分类优化算法。该算法将多重信号分类算法输出值作为初始值,并在Group Lasso算法组间计算时对目标信号进行稀疏、在组内计算时对该组信号进行平滑及阈值截断。仿真结果表明:该优化算法在中低频段可明显提高多重信号分类算法分辨率,同时改善因扫描位置与声源面位置不重合引起的聚焦性能下降问题。     

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.     

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     

循环平稳声场的声源定位研究

旋转机械产生的辐射声场具有循环平稳特性,传统的平面近场声全息技术无法准确反映其调制特性,往往在边频带上出现虚假的能量的分布.采用循环谱密度取代功率谱密度作为重建物理量,则可准确提取循环平稳声场的调制和载波信息.考虑到循环谱密度的计算量以及特征提取的准确性,提出了循环谱密度组合切片分析法,并分析了加性白噪声对重建的影响.仿真分析及实验结果表明,此方法有较强的噪声抑制能力,全息重建的结果可准确反映声源的位置.     

Sound source localization in real sound fields based on empirical statistics of interaural parameters

The role of temporal fluctuations and systematic variations of interaural parameters in localization of sound sources in spatially distributed, nonstationary noise conditions was investigated. For this, Bayesian estimation was applied to interaural parameters calculated with physiologically plausible time and frequency resolution. Probability density functions (PDFs) of the interaural level differences (ILDs) and phase differences (IPDs) were estimated by measuring histograms for a directional sound source perturbed by several types of interfering noise at signal-to-noise ratios (SNRs) between -5 and +30 dB. A moment analysis of the PDFs reveals that the expected values shift and the standard deviations increase considerably with decreasing SNR, and that the PDFs have non-Gaussian shape at medium SNRs. A d' analysis of the PDFs indicates that elevation discrimination is possible even at low SNRs in the median plane by integrating information across frequency. Absolute sound localization was simulated by a Bayesian maximum a posteriori (MAP) procedure. The simulation is based on frequency integration of broadly tuned "detectors." Confusion patterns of real and estimated sound source directions are similar to those of human listeners. The results indicate that robust processing strategies are needed to exploit interaural parameters successfully in noise conditions due to their strong temporal fluctuations.     

Impact of unilateral hearing loss on sound localization

The impact of unilateral hearing loss on the localization of horizontal plane sound sources ipsilateral and contralateral to the side of the unimpaired ear was examined. Normal-hearing listeners judged the direction of six loudspeakers, separated by 30° and arrayed frontally or laterally on the right side with the right or left ear occluded. The benefit of massed practice over three sequential days was assessed. For the frontal loudspeaker array, azimuthal discrimination on the occluded side was poor but only 30% of sounds were perceived to come from the unoccluded side. For the right lateral array, when the ipsilateral ear was unoccluded, front and back were rarely confused. Accuracy mainly decreased for speakers close to the midline axis, front and back. When the contralateral ear was unoccluded responses were biased toward the rearmost speaker. Practice did not improve performance. The findings were discussed within the context of military operations. They support the need for job-specific hearing standards.     

基于定量分析的气固两相流声速模型综述

声波法是多相流测量的一种有效方法,可靠的声速模型是其中时差法等测量的理论基础。该文系统总结了国内外不同学者所建立的气固两相声速模型。为直观地呈现其差异、发展与局限,该文以颗粒相浓度的影响为例,采用模拟计算和实验测量相结合的方法对其中不同典型声速模型进行了定量分析与比较。结果表明,传统的拟均相介质模型声速随颗粒浓度变化,与在其基础上考虑声波特性的改进模型,以及分别基于相间耦合和颗粒散射的声速模型差异均较大,且显著偏离实验测量值。其主要源于传统的拟均相介质模型未考虑两相间相对运动的影响;而考虑相间相互作用的声速模型间虽有不同差异,但均与实验测量值较为接近,根据稀相区实验研究,所得声速值均较实验值偏小。