一种利用分布式传声器阵列的声源三维定位方法

为了提高噪声和混响条件下分布式传声器阵列进行声源定位的性能,提出一种利用空间稀疏性和压缩感知原理的声源三维定位方法。该方法首先通过两次离散余弦变换方式提取出声音信号特征,并用该特征来构建稀疏定位模型,以便能够综合利用语音信号的短时和长时特性,同时降低模型维数;然后利用在线字典学习技术动态调整字典,克服稀疏模型与实际信号之间的失配问题,增强稀疏定位模型的鲁棒性;进而提出一种改进的平滑l₀范数稀疏重构算法来进行声源位置解算,以提高低信噪比条件下的重构精度。仿真结果表明该方法不仅可以实现多目标定位,而且具有较强的抗噪声和抗混响能力.      

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.     

Robust tracking of moving sound source using multiple model Kalman filter

In this paper a novel method for tracking an active speaker in a noisy and reverberant environment by means of a spatially distributed microphone array is presented. Firstly, a sound source localization algorithm based on time delays of arrival (TDOA) in microphone pairs provides observed position estimates. Then these remarkably noisy estimates are filtered by a multiple model Kalman filter (MMKF) in order to obtain a smoothed trajectory of the speaker’s movement. Compared with the traditional Kalman filter (KF), simulated results prove the MMKF is more robust and effective in noisy environments.     

一种强噪声背景下微弱超声信号提取方法研究

为解决在强噪声背景下获取超声信号的难题,基于粒子群优化算法和稀疏分解理论提出一种强噪声背景下微弱超声信号提取方法.该方法将降噪问题转换为在无穷大参数集上对函数进行优化的问题,首先以稀疏分解理论和超声信号的结构特点为依据构建了粒子群优化算法运行所需要的目标函数及去噪后信号的重构函数,从而将粒子群优化算法和超声信号降噪联系在一起;然后根据粒子群优化算法可以在连续参数空间寻优的特点建立了用于匹配超声信号的连续超完备字典,并采用改进的自适应粒子群优化算法在该字典中对目标函数进行优化;最后根据对目标函数在字典上的优化结果确定最优原子,并利用最优原子按照重构函数重构出降噪后的超声信号.通过对仿真超声信号和实测超声信号的处理,结果表明本文提出的方法可以有效提取信噪比低至-4 dB的强噪声背景下的微弱超声信号,且和基于自适应阈值的小波方法相比本文方法表现出更好的降噪性能.     

基于压缩感知的矢量阵聚焦定位方法

本文针对噪声源近场定位识别问题,利用声源分布在空间域具有稀疏性,在压缩感知理论框架下建立了新体系下的矢量阵聚焦波束形成方法,用于解决同频相干声源的定位识别问题.新方法可在小快拍下准确获得噪声源的空间位置,且不损失对噪声源贡献相对大小的评价能力.通过详细的理论推导、仿真分析和试验验证,证明了基于压缩感知的矢量阵聚焦定位新方法本质上实现了l1范数正则化求解下的波形恢复和空间谱估计,因此具有较高的定位精度,较强的相干声源分辨能力、准确的声源贡献相对大小评价能力以及较高的背景压制能力,可应用于水下复杂噪声源的定位识别.     

Accelerated steered response power method for sound source localization using orthogonal linear array

In microphone arrays application, it is a difficult task to accurately and fast localize sound source in a noisy, reverberant environment. In order to solve this problem, many approaches have been presented. Among them, the steered response power-phase transform weighted (SRP–PHAT) source localization algorithm has been proved robust. However, SRP–PHAT requires high computation cost for searching a large location space. To overcome this shortcoming, an improved SRP–PHAT will be presented that reduces a two-dimension searching space into a couple of one-dimension ones by using an orthogonal linear array. In this method, the parameters of direction of arrival (DOA) are separated. The main computation can be carried out independently in two one-dimension spaces, thus the computational load will be greatly cut down. Simulations show that there is no loss in accuracy in the proposed method.     

基于稀疏表示和特征加权的离格双耳声源定位*

基于头相关传递函数数据库的传统双耳声源定位方法的定位角度往往被限定在头相关传递函数数据库的离散测量点上。当头相关传递函数数据库的测量方位角间隔较大时,这类算法的性能会显著下降,这就是典型的离格问题。该文提出了基于加权宽带稀疏贝叶斯学习的离格双耳声源定位算法。首先该算法建立离格双耳信号的稀疏表示模型,然后利用双耳相干与扩散能量比特征对各个频点进行加权以降低噪声和混响的影响,最后通过加权宽带稀疏贝叶斯学习方法估计离格声源的方位角。实验结果表明,该算法在各种复杂的声学环境下都有着较高的定位精度和鲁棒性,特别是提高了离格条件下的声源定位性能。     

分布式压缩感知麦克风阵列多声源方位估计*

麦克风阵列已被广泛应用于音/视频会议等人机交互领域中时,多声源应用场景对声源方位估计性能提出了更高的要求。压缩感知(CS)声源定位算法将声源定位问题转化为信号的稀疏重构问题,相比传统的定位算法如相位变换加权（SRP-PHAT）和时延累加定位（DS）能够获得较高的定位性能,但多声源的存在一定程度上降低了稀疏程度,影响了CS重构性能。考虑到传统的CS定位算法并未利用多个连续语音帧之间声源空间向量的共同稀疏性,提出采用分布式压缩感知（DCS）理论以改善多声源的稀疏恢复估计的性能。仿真和实验结果表明,相比于传统定位算法和CS-OMP算法,DCS-SOMP算法在不同信噪比和不同声源强度的环境中,对多声源的方位估计都具有更好的定位性能和定位稳健性。     

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。      

混响环境下目标声场稀疏分解多域声场重放方法*

对于封闭空间内的声场重放,传统的多点声压匹配方法（Cov-PM）直接依据在目标声场测得的声压直接使用最小二乘来计算扬声器权重。然而这种方法要求较多的目标声场采样点以实现足够的精度。对于一类特殊的声场景,也就是目标声场是由少量声源辐射产生时,提出一种对目标声场稀疏分解的方法来进行混响环境下多域声场重构以降低对目标声场采样数量的要求。本文给出基于目标声场等效源稀疏分解多域重构方法（Sparse-ESM）理论推导,通过数值计算以及实验测试两种方式对比所提方法与最小二乘等效源分解方法以及Cov-PM的声场重放性能。数值结果表明,在600Hz 以上的频段,Sparse-ESM 方法的重构误差性能提升明显。实验结果也得出了与数值计算相同的结论。同时,还通过数值计算和实验测试两种方法证明了当目标声场声源方向波动时,Sparse-ESM 仍然可以保持与其余两类方法相近的声对比度同时实现较高的亮区重构精度。     

基于字典学习和稀疏表示的单通道语音增强算法综述*

如何从带噪语音信号中恢复出干净的语音信号一直都是信号处理领域的热点问题。近年来研究者相继提出了一些基于字典学习和稀疏表示的单通道语音增强算法,这些算法利用语音信号在时频域上的稀疏特性,通过学习训练数据样本的结构特征和规律来构造相应的字典,再对带噪语音信号进行投影以估计出干净语音信号。针对训练样本与测试数据不匹配的情况,有监督类的非负矩阵分解方法与基于统计模型的传统语音增强方法相结合,在增强阶段对语音字典和噪声字典进行更新,从而估计出干净语音信号。本文首先介绍了单通道情况下语音增强的信号模型,然后对4种典型的增强方法进行了阐述,最后对未来可能的研究热点进行了展望。     

冲击声的稀疏特征提取及声源类型识别

针对低信噪比下声源材料类型的细分任务,将稀疏表达用于冲击声信号的声源类型识别,提取的稀疏特征相比传统的MFCC特征有效改善了识别性能。分别基于3种预定义词典和一组根据训练信号学习的词典,利用正交匹配追踪(OMP)方法对录制冲击声进行稀疏表达,提取稀疏特征用于不同信噪比下冲击声信号的声源辨识,并与MFCC特征进行比较。对包含12类材料的冲击声数据库的分类结果显示,在几乎所有情况下,稀疏特征比MFCC特征具有更好的识别效果。特别是在信噪比较低的情况下,稀疏特征具有更好的抗噪性能。      

球谐域自适应混响抵消与声源定位算法

提出了一种基于球谐域的自适应混响抵消与声源定位算法,该方法通过去混响处理改善语音质量,并提高球谐域定位算法在混响环境下的定位性能。推导了基于多通道线性预测的自适应混响抵消算法在球谐域的表达式,针对刚球模型提出分阶处理的去混响方法,并对去混响后的信号进行波达方向估计。采用32元球阵的仿真结果表明,相比于球谐域不分阶去混响方法,该方法最大可减少约2/3的运算量,同时语音PESQ得分及SRMR均显著提高。利用实验数据对算法性能进行测试,实验结果验证了该方法在实际声学环境中去混响和声源定位的有效性。      

混响声场条件下水声互易传递函数的测量及应用

对不规则的混响声场情况的互易测量方法进行了理论研究,对相关测量误差提出了修正方法。通过测量结果的空间平均和混响环境对声能密度影响的评估,修正了声源体积速度的计算值,并在水中进行了试验验证。经修正后的互易测量结果与正向实测结果基本一致。表明混响声场不影响互易原理有效性的成立,但影响了声源体积速度的计算,进而影响了应用互易原理对传递函数的测量。提出的修正方法在不规则的混响声场情况下简单有效,结论为今后基于互易原理的工程应用提供了参考和依据。      

稀疏低秩噪声模型下无监督实时单通道语音增强算法

针对现有基于字典学习的增强算法需要先验信息、不易实时处理的问题,提出一种便于实时处理的无监督的单通道语音增强算法。首先,该算法将无监督条件下背景噪声的建模问题转化为带噪语音幅度谱的稀疏低秩噪声分解;然后,采用增量非负子空间方法对背景噪声进行在线字典学习,获得能够体现背景噪声时变特性的自适应噪声字典;最后,利用所得的噪声字典,采用易于实时处理的逐帧迭代方式,对带噪语音进行处理。实验结果表明:相较于多带谱减法和基于低秩稀疏矩阵分解的增强算法,所提算法在噪声抑制方面的性能尤为显著,在多项性能评价指标上,均表现出更好的结果。      

Application and measurement of underwater acoustic reciprocity transfer functions in reverberant sound field

The reciprocity measurement theory in anomalous reverberant sound fields was investigated.An improved method Was proposed due to the interrelated errors.The source volume velocity Was corrected by spatial average of measurement results and evaluation of the reverberant sound field influence on acoustic energy density.The result was validated in underwater experiment,corrected reciprocity measurement results were almost the same as direct measurement results.It indicates that reverberant sound field does not affect the validitv of the principle,but influences the obtainment of source volume velocity,then influences the measurement of transfer functions with the principle.The proposed method is simple and effective in anomalous reverberant sound fields.The study mav be valuable for the applications which are based on the principle.     

Measuring sound scattering coefficients of uneven surfaces in a reverberant workplace – Principle and validation of the method

In workplaces, wall facings are often based on periodic or aperiodic sound scattering surfaces. It is necessary to develop acoustic characterization methods for these kinds of walls to predict the acoustic pressure cartography in the room in order to improve the acoustical treatment. However, this characterization is quite difficult because of the partially reverberant conditions. We developed a measurement system which determines in situ the sound scattering coefficients of relief surfaces. The measurement method, originally operating in free-field conditions, was adapted for indoor use. To overcome problems of parasite echoes coming from reverberation and from noisy sources present on the site, we developed a dedicated emission/reception system. An acoustic antenna with constant directivity over the full frequency range allows spatial filtering of the parasite echoes and an impulsive sound source enables the use of a broad temporal window, resulting in adequate time separation of the different signals received by the antenna. Measurements of the sound scattering coefficient of a corrugated panel were carried out for several incidence angles in free-field and in a noisy workshop and allowed the in situ validation of this system.     

Effect of source spectrum on sound localization in an everyday reverberant room

Two experiments explored how frequency content impacts sound localization for sounds containing reverberant energy. Virtual sound sources from thirteen lateral angles and four distances were simulated in the frontal horizontal plane using binaural room impulse responses measured in an everyday office. Experiment 1 compared localization judgments for one-octave-wide noise centered at either 750 Hz (low) or 6000 Hz (high). For both band-limited noises, perceived lateral angle varied monotonically with source angle. For frontal sources, perceived locations were similar for low- and high-frequency noise; however, for lateral sources, localization was less accurate for low-frequency noise than for high-frequency noise. With increasing source distance, judgments of both noises became more biased toward the median plane, an effect that was greater for low-frequency noise than for high-frequency noise. In Experiment 2, simultaneous presentation of low- and high-frequency noises yielded performance that was less accurate than that for high-frequency noise, but equal to or better than for low-frequency noise. Results suggest that listeners perceptually weight low-frequency information heavily, even in reverberant conditions where high-frequency stimuli are localized more accurately. These findings show that listeners do not always optimally adjust how localization cues are integrated over frequency in reverberant settings.     

合成孔径压缩感知超声成像中的高效能稀疏字典设计

针对目前常见的稀疏字典缺乏针对性,在合成孔径医学超声成像中的应用效果不佳,难以在低压缩率下保证重构图像质量的问题,本文设计了一种高效能的稀疏字典。根据超声回波信号是由发射脉冲信号经过不同延时衰减后叠加的特点,利用发射脉冲作为基函数构造稀疏字典,回波信号在该稀疏字典确定的变换域中具备很好的稀疏性,理论上能使其稀疏表示系数的稀疏度等于超声阵元接收到的反射回波数。通过FieldⅡ对简单点目标和复杂目标的仿真结果表明:在相同的重构算法和压缩率下该稀疏字典重构的平均绝对误差明显小于常见的稀疏字典,其值仅为DWT的几分之一,DFT和DCT的几十分之一,能让回波信号以更低的压缩率实现相同的恢复效果。本文最后使用体模的实际采集数据对算法的实际效果进行检测,实验结果也与仿真结果基本一致。基于该稀疏字典的压缩感知算法可以进一步减少合成孔径成像所需存储的数据量、降低系统的复杂度。      

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。