浅海波导环境失配下的协方差矩阵拟合稳健匹配场声源功率估计

实际浅海波导中环境噪声为相干噪声,最小方差匹配场声源功率估计方法能在相干噪声背景下准确估计声源辐射功率,但该方法受环境不确定性影响较大;此外,由于最小方差匹配场声源功率估计方法使用信号幅度作为中间量估计声源功率,信号幅度估计误差会二次放大并传递到声源功率估计结果中。本文提出一种协方差矩阵拟合稳健最小方差匹配场声源功率估计方法,该方法引入信道传递函数不确定集,结合协方差矩阵拟合思想将声源功率估计问题建模为在信道传递函数不确定集约束下对函数取极值的问题,使用Lagrange乘子法求解该问题得到信道传递函数估计值和声源辐射功率估计值。环境失配影响声源辐射功率估计性能的根本原因在于信道传递函数偏差较大,协方差矩阵拟合稳健匹配场声源功率估计方法有效减小了环境失配时信道传递函数的偏差,从而显著提升环境失配稳健性。此外,该方法使用权值直接估计声源功率,无需使用信号幅度作为中间量,避免了估计误差的传递。仿真验证了协方差矩阵拟合稳健匹配场声源功率估计方法的环境失配稳健性。     

一种鲁棒性的最小方差无失真响应波束形成算法及其应用

理论上,自适应波束形成方法要比不依赖于输入数据的常规波束形成方法有更好的目标参数估计能力和干扰抑制能力.但在实际水声环境中,声传播模型、接收阵阵列流形以及信号统计特征等因素往往与实际情况存在一定的差异,导致传统的自适应波束形成方法性能下降.因此,提高自适应波束形成方法对上述因素的鲁棒性变得越来越重要.本文基于最差条件最优化的思想,改进MVDR(最小方差无失真响应)方法的约束条件提出了一种鲁棒性最小方差无失真响应自适应波束形成算法(R-MVDR),并对输入数据协方差矩阵和方向向量存在不确定性的情况进行了性能分析,推导给出了波束形成的加权向量和空间谱估计表达式,最后通过海上实验数据进行了验证.结果证明本文提出的算法在实际环境中有更好的方位分辨能力和干扰抑制能力.     

波束空间能量约束的稳健自适应波束形成

为改善波束形成器在一定失配条件下的工作性能,提出了一种稳健自适应波束形成方法。该方法通过设计信号在波束空间能量分布的约束条件,利用阵列数据协方差矩阵,结合迭代二阶锥规化方法,实现信号驾驶向量的估计,最后利用该估计得到最小方差无失真响应波束形成器。该方法在较恶劣的失配条件下也可以自适应地准确估计信号驾驶向量并保证其成分无失真地通过,使系统的稳健性得到改善。数据仿真结果表明,该方法在理想条件下与成熟的对角加载法性能相当,在失配条件下则优于后者。      

快速收敛最小方差无畸变响应算法研究及应用

常规最小方差无畸变响应(MVDR)自适应波束形成是一种高分辨窄带波束形成器,它是利用实际声场的窄带互谱密度矩阵(CSDM)估计出自适应波束形成权向量。在实际应用中,MVDR算法需要较长的观测时间估计协方差矩阵,不利于对高速运动目标进行定位;对于宽带目标信号,MVDR算法需要对每一个CSDM进行求逆运算,计算量较大;在相干源条件下,目标信号之间会发生"对消"现象,MVDR算法性能急剧恶化。本文提出了基于子带子阵处理的快速收敛MVDR自适应波束形成方法。首先将全频带划分成一组子带,将接收线阵划分成一组子阵,然后对每一子带计算降维的驾驶协方差矩阵(STCM),从而得到快速收敛MVDR自适应波束形成的权值和空间谱估计结果。同时采用双向空间平滑方法对相干源进行MVDR空间谱估计。仿真和海试数据处理结果表明该算法在保证高分辨力的同时,具有瞬时收敛的性能,双向空间平滑技术具有良好的解相干性能。      

频域盲源分离与波束形成结合抑制方向性强干扰方法

针对方向性强干扰严重影响无源声呐弱目标检测的问题,提出了频域盲源分离与波束形成结合的干扰抑制方法:以子带分解的方法实现宽带干扰抑制。对每个子带进行频域盲源分离,并估计出各分离信号的方位,将与给定强干扰方位匹配的分离信号置零,利用估计的解混矩阵和处理后的分离信号重构回阵元域信号并进行波束形成实现目标方位估计。声呐模拟器数据与海试数据验证结果表明,相对于传统零陷常规波束形成与零陷最小方差无失真响应波束形成方法有2 dB以上的增益,约6 dB的背景级降低,证明该方法在抑制方向性强干扰方面是有效的。      

频域单快拍压缩感知目标方位估计和信号恢复方法

方位估计和信号恢复分别是水下目标定位、跟踪和目标识别的前提。提出了一种阵列频域单快拍压缩感知的水下目标方位估计和信号恢复方法。首先将阵列接收数据变换到频域,取频域单快拍数据作为压缩感知的测量值,然后根据频域快拍对应的频率、搜索方位和阵列流形构造过完备的阵列流形矩阵作为压缩感知的感知矩阵,最后通过基追踪算法估计搜索方位上目标信号和功率,实现DOA估计与信号恢复。宽带仿真实验数据验证结果表明,同等条件下完成同样的目标方位分辨,提出的方法比最小方差无失真响应方法要求的阵元数和快拍数较少,要求的信噪比更低,恢复的目标信号更加准确,波形相关系数达到89%以上。海上实验数据处理结果表明,目标检测能力优于最小方差无失真响应方法,证明该方法可以适用于实际声呐系统。      

特征空间和符号相干系数融合的最小方差超声波束形成

为了提高医学超声成像的空间分辨率,提出一种融合了特征空间最小方差与符号相干系数的波束形成方法。首先利用最小方差法计算回波数据的协方差矩阵和加权向量;然后对协方差矩阵进行特征分解得到信号子空间,并将加权向量投影到该空间上;最后计算符号相干系数,用于优化特征空间法得到的回波信号,最终获得超声成像数据。为验证算法的有效性,对医学超声成像中常用的点目标、斑目标进行仿真,对点目标仿体和人体颈动脉组织进行超声成像实验。结果表明:所提出的方法在分辨率、对比度以及稳健性等方面都优于传统的延时叠加算法、最小方差算法、特征空间最小方差法以及特征空间与相干系数融合的方法。     

稳健的子带子阵级导向最小方差波束形成算法

导向最小方差(STMV)波束形成是一种利用导向协方差矩阵获得自适应权值的方法,具有快速收敛特性.常规的稳健导向最小方差(RSTMV)波束形成算法在处理宽频带信号时,性能下降明显.为了改善算法的性能,结合频域子带划分和空域子阵划分技术,提出一种多子带不确定集独立约束的稳健子阵级STMV波束形成算法。通过频域子带划分可对不同子带的导向向量误差范数边界进行约束,计算出各子带对应的对角加载量,得到稳健的子带级最小方差波束形成算法权向量;同时采用子阵技术进行降维处理,可进一步增加划分子带的数目,从而提高算法的性能并有效降低计算复杂度,最终得到一种稳健的子带子阵级STMV波束形成算法。理论分析和仿真结果表明,在阵列导向向量存在误差的情况下,该算法在干扰方向形成的零陷最深,且零陷波束宽度最窄,输出信噪比接近理论值,因此性能最佳.实际海试数据处理表明,在强干扰目标存在时,弱目标输出信干噪比较RSTMV算法可提高4 dB,较常规波束形成可提高10 dB,在角度分辨力和算法复杂度方面得到有效改善,同时可以保证目标功率无失真输出。      

单矢量水听器稀疏近似最小方差方位估计算法

针对单矢量水听器海上目标探测问题,利用稀疏近似最小方差(Sparse Asymptotic Minimum Variance,SAMV)算法进行目标方位估计,该算法利用单矢量水听器自身具有阵列流形的特点,将整个扫描空间离散化,目标方位分布于某一离散方向位置上,利用空间信号的稀疏性可提高目标方位估计性能。仿真结果表明,SAMV算法在各信噪比条件下方位估计噪声背景级明显优于常规波束形成(Conventional Beam Forming,CBF)算法和最小方差无失真响应(Minimum Variance Distortionless Response,MVDR)算法,当信噪比大于0dB时,该算法测向结果均方根误差小于2°,且SAMV算法具有更好的空间方位分辨能力。消声水池和海上声学浮标海上试验数据处理结果表明,SAMV算法给出了噪声背景级更低的目标方位历程图,有效验证了SAMV算法对海上目标的探测性能及其有效性。      

特征空间和符号相干系数融合的最小方差超声波束形成

为了提高医学超声成像的空间分辨率,提出一种融合了特征空间最小方差与符号相干系数的波束形成方法。首先利用最小方差法计算回波数据的协方差矩阵和加权向量;然后对协方差矩阵进行特征分解得到信号子空间,并将加权向量投影到该空间上;最后计算符号相干系数,用于优化特征空间法得到的回波信号,最终获得超声成像数据。为验证算法的有效性,对医学超声成像中常用的点目标、斑目标进行仿真,对点目标仿体和人体颈动脉组织进行超声成像实验。结果表明:所提出的方法在分辨率、对比度以及稳健性等方面都优于传统的延时叠加算法、最小方差算法、特征空间最小方差法以及特征空间与相干系数融合的方法。      

Research and application of fast-convergent minimum variance distortionless response algorithm

The conventional MVDR adaptive beamformer is a high-resolution narrowband beamformer which estimates the optimal beamforming weights using narrowband CSDM of real acoustic field.In practical applications,MVDR algorithm needs long observation time to estimate the covariance matrix.This inherent property makes it difficult to localize fast-moving targets.For wideband signals,MVDR algorithm needs inverting every CSDM which increases the computational demands.For correlated sources,the performance of MVDR will degrade dramatically because the signals will cancel each other.A fast-convergent MVDR algorithm based on subband subarray processing is proposed.The full frequency band is divided into sets of subbands and the line array is divided into sets of subarrays.For every subband the STCM of reduced dimensions is calculated.Then adaptive beamforming weight of fast-convergent MVDR algorithm and spatial spectrum estimation are obtained.At the same time,spatial spectrum estimation can be made for correlated sources using the two-sided spatial smoothing method. Results of simulation and trial data show that the proposed method has high-resolution and near-instantaneous convergence property,two-sided spatial smoothing has satisfactory validity of decorrelation.     

MASS-RAB: Robust adaptive beamforming for general-rank signal models via matched spatial spectrum processing

The wavefront of acoustic signal suffers from fast fluctuation after a long distance propagation in a random and inhomogeneous ocean channel, which makes the rank of the covariance matrix for the desired signal (signal of interest) remarkably higher than one. Consequently, the assumption of rank-one point signal model for existing adaptive beamforming algorithms is no longer suitable. In this paper, a matched spatial spectrum processing based robust adaptive beamforming (MASS-RAB) algorithm is presented for general-rank signal models. First, the interference-plus-noise covariance matrix and the desired signal covariance matrix are reconstructed using the matched spatial spectrum processing method. Second, the weight vector is directly calculated using these reconstructed covariance matrices for the minimum variance distortionless response (MVDR) algorithm, which is developed for the general-rank signal models. Due to covariance matrix reconstruction, the MASS-RAB algorithm is more robust than those methods relying on the sample covariance matrix. The cases of the rank-one point signal model and the full-rank non-point signal model are considered by several numerical examples. Experimental results have demonstrated the superiority of the proposed MASS-RAB method.     

An improved TOA estimation algorithm based on denoised MVDR for B5G positioning

This paper proposes an improved minimum variance distortionless response (MVDR) based TOA estimation algorithm for 5G NR signals under multipath environments. The proposed algorithm achieves high resolution by exploiting a large number of subcarriers of 5G signals and reduces the dimension of the covariance matrix involved in MVDR substantially by utilizing a novel smoothing scheme. Since MVDR requires a relatively high signal-to-noise ratio (SNR), a denoising method is used to improve the TOA estimation performance. Simulation results show that the proposed algorithm achieves much higher resolution than the Bartlett beamformer (BF) and the TOA estimation accuracy remains high over a wide range of SNRs.     

Localization of multiple acoustic sources with small arrays using a coherence test

Direction finding of more sources than sensors is appealing in situations with small sensor arrays. Potential applications include surveillance, teleconferencing, and auditory scene analysis for hearing aids. A new technique for time-frequency-sparse sources, such as speech and vehicle sounds, uses a coherence test to identify low-rank time-frequency bins. These low-rank bins are processed in one of two ways: (1) narrowband spatial spectrum estimation at each bin followed by summation of directional spectra across time and frequency or (2) clustering low-rank covariance matrices, averaging covariance matrices within clusters, and narrowband spatial spectrum estimation of each cluster. Experimental results with omnidirectional microphones and colocated directional microphones demonstrate the algorithm's ability to localize 3-5 simultaneous speech sources over 4 s with 2-3 microphones to less than 1 degree of error, and the ability to localize simultaneously two moving military vehicles and small arms gunfire.     

Adaptive microphone array for unknown desired speaker's transfer function

The main drawback of minimum variance distortionless response (MVDR) beamformer is the cancellation of the desired speech signal and its degradation in multi-path wave propagation environment. To make the adaptive algorithm robust against room reverberation and to prevent desired signal cancellation an estimation of unknown desired speaker's transfer function was proposed. The estimation is based on the signal and the interference covariance matrices. The estimated transfer function is then applied to the MVDR beamformer. The proposed algorithm was tested on a simulated room with reverberation. The results showed better quality of the restored speech compared to some typical adaptive algorithms.     

Acoustic and linguistic factors in the perception of bandpass-filtered speech

Speech can remain intelligible for listeners with normal hearing when processed by narrow bandpass filters that transmit only a small fraction of the audible spectrum. Two experiments investigated the basis for the high intelligibility of narrowband speech. Experiment 1 confirmed reports that everyday English sentences can be recognized accurately (82%-98% words correct) when filtered at center frequencies of 1500, 2100, and 3000 Hz. However, narrowband low predictability (LP) sentences were less accurately recognized than high predictability (HP) sentences (20% lower scores), and excised narrowband words were even less intelligible than LP sentences (a further 23% drop). While experiment 1 revealed similar levels of performance for narrowband and broadband sentences at conversational speech levels, experiment 2 showed that speech reception thresholds were substantially (>30 dB) poorer for narrowband sentences. One explanation for this increased disparity between narrowband and broadband speech at threshold (compared to conversational speech levels) is that spectral components in the sloping transition bands of the filters provide important cues for the recognition of narrowband speech, but these components become inaudible as the signal level is reduced. Experiment 2 also showed that performance was degraded by the introduction of a speech masker (a single competing talker). The elevation in threshold was similar for narrowband and broadband speech (11 dB, on average), but because the narrowband sentences required considerably higher sound levels to reach their thresholds in quiet compared to broadband sentences, their target-to-masker ratios were very different (+23 dB for narrowband sentences and -12 dB for broadband sentences). As in experiment 1, performance was better for HP than LP sentences. The LP-HP difference was larger for narrowband than broadband sentences, suggesting that context provides greater benefits when speech is distorted by narrow bandpass filtering.     

波达方向初始化空间混合概率模型的语音增强

提出了波达方向初始化空间混合概率模型的语音增强算法.通过声源定位估计出声源波达方向,再根据此计算相对传递函数,进而构造空间协方差矩阵来初始化空间混合概率模型.论证了相对传递函数在作为模型参数中语音协方差矩阵的主特征向量时,空间混合概率模型对应的概率分布可达到最大值,进而使期望最大化算法在迭代时更易收敛,以得到期望的掩蔽...     

窄带语音带宽扩展算法研究

为了降低谱失真,提出了一种基于隐马尔科夫模型的窄带语音带宽扩展算法。首先,算法选取与宽带谱包络互信息大的参数构成特征矢量,并利用隐马尔可夫状态和过去观察特征矢量的联合先验概率估计条件后验概率。其次,以条件后验概率为基础,算法结合贝叶斯条件参数估计法和最小均方差准则估计宽带谱包络。针对宽带激励信号估计,基于信号高频和低频的谐波相关性,提出了一种中频激励扩展算法。实验结果表明,与传统的基于隐马尔可夫模型的带宽扩展算法相比,本文算法可降低0.187 dB的平均谱失真,将谱失真大于10 dB的语音帧减少了34.3%。      

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统阵元域估计方法低了一个数量级。并且在大部分声场环境中,尤其在低信噪比情况下,所提方法的语音质量感知评估测度得分、分段信噪比和短时客观可懂度明显高于阵元域方法,三者最高分别可提升1.31 dB,4.54 dB和35%。另外,实际声学环境的测量实验也验证了所提方法在不影响可懂度的条件下比传统阵元域方法具备更高的降噪量。