GSVD-based optimal filtering for single and multimicrophone speech enhancement

A generalized singular value decomposition (GSVD) based algorithm is proposed for enhancing multimicrophone speech signals degraded by additive colored noise. This GSVD-based multimicrophone algorithm can be considered to be an extension of the single-microphone signal subspace algorithms for enhancing noisy speech signals and amounts to a specific optimal filtering problem when the desired response signal cannot be observed. The optimal filter can be written as a function of the generalized singular vectors and singular values of a speech and noise data matrix. A number of symmetry properties are derived for the single-microphone and multimicrophone optimal filter, which are valid for the white noise case as well as for the colored noise case. In addition, the averaging step of some single-microphone signal subspace algorithms is examined, leading to the conclusion that this averaging operation is unnecessary and even suboptimal. For simple situations, where we consider localized sources and no multipath propagation, the GSVD-based optimal filtering technique exhibits the spatial directivity pattern of a beamformer. When comparing the noise reduction performance for realistic situations, simulations show that the GSVD-based optimal filtering technique has a better performance than standard fixed and adaptive beamforming techniques for all reverberation times and that it is more robust to deviations from the nominal situation, as, e.g., encountered in uncalibrated microphone arrays.     

四阶累积量稀疏表示的DOA估计方法

针对现有稀疏重构DOA估计算法不能抑制噪声项以及在高斯色噪声背景下不再适用的问题,本文提出了基于四阶累积量稀疏重构的DOA估计方法。首先,利用接收数据的四阶累积量构建了稀疏表示模型,该模型抑制了噪声项;其次对四阶累计量矩阵进行奇异值分解,化简了稀疏表示模型,通过奇异值分解,不仅减小了数据规模,而且进一步抑制了噪声。对于稀疏表示模型的求解,先利用信号子空间与噪声子空间的正交特性选取权值矢量,然后利用加权l1范数法对模型求解实现DOA估计。理论分析和仿真实验表明本文算法在高斯白噪声和色噪声背景下均适用;能够处理非相干和相干信号,且在低信噪比条件下,对相干信号有更高的估计精度;较之同类的稀疏重构算法,本文算法具有较低的算法复杂度和更高的角度分辨力。      

改进的基于信号子空间的多通道语音增强算法

通过同时对角化麦克风阵列接收信号中语音信号和噪声信号的全局协方差矩阵,本文改进了一种基于信号子空间分解的多通道语音增强算法.该算法不依赖任何信号模型且无需对噪声信号的统计特性进行任何先验假定,它弥补了原始算法只限于白噪声背景下语音增强的不足,实现了色噪声背景下语音信号的最优估计.仿真结果表明本文算法在主观和客观测试中都具有良好的语音增强效果.     

Noise covariance incorporated MEG-MUSIC algorithm: a method formultiple-dipole estimation tolerant of the influence of background brainactivity

This paper proposes a method of localizing multiple current dipoles from spatio-temporal biomagnetic data. The method is based on the multiple signal classification (MUSIC) algorithm and is tolerant of the influence of background brain activity. In this method, the noise covariance matrix is estimated using a portion of the data that contains noise, but does not contain any signal information. Then, a modified noise subspace projector is formed using the generalized eigenvectors of the noise and measured-data covariance matrices. The MUSIC localizer is calculated using this noise subspace projector and the noise covariance matrix. The results from a computer simulation have verified the effectiveness of the method. The method was then applied to source estimation for auditory-evoked fields elicited by syllable speech sounds. The results strongly suggest the method's effectiveness in removing the influence of background activity     

Subspace averaging of steady-state visual evoked potentials

A new algorithm for doing signal averaging of steady-state visual evoked potentials (VEP's) is described. The subspace average is obtained by finding the orthogonal projection of the VEP measurement vector onto the signal subspace, which is based on a sinusoidal VEP signal model. The subspace average is seen to out-perform the conventional average using a new signal-to-noise-ratio-based performance measure on simulated and actual VEP data.     

A two-stage algorithm for MIMO blind deconvolution of nonstationarycolored signals

A new two-stage algorithm is proposed for the deconvolution of multi-input multi-output (MIMO) systems with colored input signals. While many blind deconvolution algorithms in the literature utilize high order statistics of the output signal for white input signals, the additional information contained in colored input signals allows the design of second-order statistical algorithms. In fact, practical signal sources such as speech signals do have distinct, nonstationary, colored power spectral densities. We present a two-stage signal separation approach in which the first step utilizes a matrix pencil between output auto-correlation matrices at different delays, whereas the second stage adopts a subspace method to identify and deconvolve MIMO systems     

Modeling common dynamics in multichannel signals with applications to artifact and background removal in EEG recordings

Removing artifacts and background electroencephaloraphy (EEG) from multichannel interictal and ictal EEG has become a major research topic in EEG signal processing in recent years. We applied for this purpose a recently developed subspace-based method for modeling the common dynamics in multichannel signals. When the epileptiform activity is common in the majority of channels and the artifacts appear only in a few channels the proposed method can be used to remove the latter. The performance of the method was tested on simulated data for different noise levels. For high noise levels the method was still able to identify the common dynamics. In addition, the method was applied to real life EEG recordings containing interictal and ictal activity contaminated with muscle artifact. The muscle artifacts were removed successfully. For both the synthetic data and the analyzed real life data the results were compared with the results obtained with principal component analysis (PCA). In both cases, the proposed method performed better than PCA.     

基于人耳感知掩蔽效应的子空间语音增强算法研究

针对传统子空间语音增强算法中,因语音增强方法中去除噪声而出现的音乐噪声和失真问题,提出了一种人耳感知掩蔽效应的子空间语音增强算法,并结合频域到特征值域的变换,在Bark域内实现人耳的感知掩蔽效应的语音增强。实验结果表明,该算法在白噪声和有色噪声的背景下,与传统子空间语音增强算法相比,不仅提高了语音信号的信噪比,而且减少了语音失真和音乐噪声,提高了增强后语音的听觉质量。     

A peak preserving algorithm for the removal of colored noise from signals

We employ the simulated annealing method in order to restore signals that may contain peaks we wish to preserve but, otherwise, are smooth and buried in colored noise. The problem is formulated as a global optimization one. We propose a piecewise linear model for the noise-free signal that is appropriate for preserving peaks that are often encountered in biomedical or industrial signals. An iterative algorithm is proposed. The restored signal is used to estimate the model parameters that are subsequently used to improve the signal estimation. The algorithm stops when self-consistency has been achieved, i.e., the estimated values of the noise model parameters agree to within the accuracy of their estimation with the parameters used to restore the signal. Application of the method to simulated data with various levels of noise showed that the underlying signal can be restored sufficiently well. The algorithm is also applied to some evoked-response magneto-encephalographic data, as well as to some signals from an automatic industrial inspection problem. Our results are compared with those obtained by using the iterative conditional modes algorithm and shown to be better in terms of preserving the peaks in the signals.     

Alternate half-field stimulation technique for SSVEP-based brain-computer interfaces

A technique of half-field alternate visual stimulation, combined with differential EEG signal measurement, is applied to acquire steady-state brain-evoked signals. Taking the difference of two signals, measured with properly placed electrodes, enhances the visual-evoked potential (VEP) and suppresses the noise components. An array of different-flash-frequency light-emitting-diode (LED) pairs forms a multiple-choice table. By fixating at different LED pairs, the user communicates (by the VEP of corresponding frequency) his/her decision about the selection of the table entry.     

基于子空间旋转不变性的加性有色噪声中谐波频率的估计

噪声中的谐波恢复问题是信号处理领域的一个典型问题,在众多领域中有着广泛的应用。本文主要研究加性有色噪声中谐波频率的估计问题,提出了一种基于子空间旋转不变性的谐波频率的高分辨率估计方法。利用观测信号的自协方差函数构造了一个协方差矩阵,通过对协方差矩阵的特征值进行理论分析,结合子空间旋转不变性,得到了加性有色噪声中谐波的频率和协方差矩阵之间的一种内在联系。利用这个性质可以估计加性有色噪声中谐波的频率。本文方法对于有色噪声的模型无任何假设,而且对于噪声的分布也没有限制,对于高斯和非高斯有色噪声都适用。仿真实验验证了本文所提算法的有效性。      

基于内插阵列变换的非圆信号MUSIC算法

针对非圆信号的波达方向(DOA)估计问题,提出一种基于内插阵列变换的非圆信号MUSIC算法(VIA-NC-MUSIC算法)。利用真实阵列流型与虚拟阵列流型之间的变换矩阵,将真实协方差矩阵变换为虚拟协方差矩阵,再对虚拟协方差矩阵进行奇异值分解(SVD),利用信号子空间与噪声子空间的正交性,得出算法的空间谱函数。仿真实验表明:存在阵元位置误差的情况下,新算法通过对阵元位置校准数据进行内插阵列变换(VIA),取得与阵元位置校准的非圆信号MUSIC算法(NC-MUSIC算法)相当的估计性能,保持了高估计精度、阵列扩展能力等优点。     

基于信号子空间的改进OFDM系统信道半盲估计

本文改进了一种基于信号子空间的OFDM系统半盲信道估计算法.该算法利用基于梯度变化的变遗忘因子递归最小二乘算法(GVFF-RLS)计算接收信号的自相关矩阵.通过同时对角化接收信号中的信息信号和噪声信号的全局协方差矩阵,推导出噪声信号子空间,无需对噪声信号的统计特性进行任何先验假定.本算法弥补了原始算法在慢衰落信道下收敛慢以及只限于加性白噪声的不足,实现了色噪声背景下高效半盲信道估计.仿真结果表明本文提出的算法具有良好的性能.     

基于部分平行ADMM求解谱寻求问题的时频分析

经典的非参数谱分析方法使用滑动窗口来捕捉大多数时间序列的频谱特性,然而这种方法不能很好地应用在时间序列的时频谱是时间连续的信号上.对于一些其时频谱满足时间连续频率稀疏的非平稳信号,提出了一种利用部分平行交替方向乘子法（Alternating Direction Method of Multipliers,ADMM）求解谱寻求问题用于此类信号的时频分析方法.对一段加噪声的仿真信号和一段EEG（脑电）信号使用提出的方法进行时频分析.仿真结果表明：与短时傅里叶的分析方法相比,提出的方法不仅提高了时频谱的频率分辨率和时间分辨率,还有效抑制了噪声.最后从ADMM算法停止准则的角度说明了算法的收敛.     

一种基于信号子空间和听觉掩蔽效应的语音增强方法

提文推导了基于离散余弦变换(DCT)的子空间分解法对有色噪声背景下的语音进行增强的公式,用基于听觉掩蔽效应的感智滤波器对增强后的信号频谱进行平滑以抑制背景噪声。几种噪声背景下对增强语音的客观测试表明,本文提出的方法可以有效地减少语音信号的失真度。     

稀疏分解的加权迭代方法及其初步应用

为了在强噪声背景下提取信号,本文发展了一种加权迭代稀疏分解方法.从一个完备库中寻找观测信号的稀疏成分表达问题的目标函数,可以取残差的l-2模和稀疏成分的l-1模的加权和最小,通过分析噪声信号在多分辨小波分解下的性质,得到了二尺度小波框架下不同尺度空间的加权系数的表达式;通过分析最小l-1模问题的求解过程,提出了用两次迭代得到的信号成分的l-1模的差作为迭代的收敛条件.最后用仿真试验和真实信号验证了方法的有效性.     

Matrix-Group Algorithm via Improved Whitening Process for Extracting Statistically Independent Sources From Array Signals

This paper addresses the problem of blind separation of multiple independent sources from observed array output signals. The main contributions in this paper include an improved whitening scheme for estimation of signal subspace, a novel biquadratic contrast function for extraction of independent sources, and an efficient alterative method for joint implementation of a set of approximate diagonalization-structural matrices. Specifically, an improved whitening scheme is first developed by estimating the signal subspace jointly from a set of diagonalization-structural matrices based on the proposed cyclic maximizer of an interesting cost function. Moreover, the globally asymptotical convergence of the proposed cyclic maximizer is analyzed and proved. Next, a novel biquadratic contrast function is proposed for extracting one single independent component from a slice matrix group of any order cumulant of the array signals in the presence of temporally white noise. A fast fixed-point algorithm that is a cyclic minimizer is constructed for searching a minimum point of the proposed contrast function. The globally asymptotical convergence of the proposed fixed-point algorithm is analyzed. Then, multiple independent components are obtained by using repeatedly the proposed fixed-point algorithm for extracting one single independent component, and the orthogonality among them is achieved by the well-known QR factorization. The performance of the proposed algorithms is illustrated by simulation results and is compared with three related blind source separation algorithms     

基于协方差矩阵重构和导向矢量估计的稳健自适应波束形成

实际应用中, 当假定的与真实的期望信号导向矢量之间存在一定误差时, 波束形成器的性能会急剧下降, 特别是当期望信号功率很强的时候.为解决这个问题, 提出了一种新的算法.当信源数小于阵元数时, 干扰加噪声协方差矩阵具有稀疏性.新方法首先利用该特性重构干扰加噪声协方差矩阵并由此得到与干扰导向矢量正交的子空间, 使接收的数据通过该子空间得到只含有期望信号和噪声的混合信号, 然后,对该混合信号基于最大化输出功率原理估计期望信号导向矢量, 最后,把得到的导向矢量和正交子空间来构造阵列加权值.仿真结果表明:该算法分别在假定的期望信号导向矢量存在误差、期望信号很强和低快拍数时仍然具有良好的性能.     

A New Approach for Estimating the Number of Sources Under the Coexistence of Circular and Various Noncircular Sources

Estimation of source number is a fundamental problem of direction-of-arrival (DOA) estimation. In the problem of DOA estimation under the coexistence of circular and various noncircular signals, the source number should be estimated in order to distinguish the signal subspace from the noise subspace. Thus, a new method for source number estimation is proposed in this paper. Using the approach of k-means clustering, the projections of a one-dimensional reduced covariance matrix are divided into two categories. Then the signal subspace and the noise subspace are separated by the optimal classification boundary of those two categories so as to obtain the equivalent source number. Simulation results show that the proposed method has relatively better performance even in low SNR or in a colored noise environment.     

Noise Removal From Hyperspectral Images by Multidimensional Filtering

A generalized multidimensional Wiener filter for denoising is adapted to hyperspectral images (HSIs). Commonly, multidimensional data filtering is based on data vectorization or matricization. Few new approaches have been proposed to deal with multidimensional data. Multidimensional Wiener filtering (MWF) is one of these techniques. It considers a multidimensional data set as a third-order tensor. It also relies on the separability between a signal subspace and a noise subspace. Using multilinear algebra, MWF needs to flatten the tensor. However, flattening is always orthogonally performed, which may not be adapted to data. In fact, as a Tucker-based filtering, MWF only considers the useful signal subspace. When the signal subspace and the noise subspace are very close, it is difficult to extract all the useful information. This may lead to artifacts and loss of spatial resolution in the restored HSI. Our proposed method estimates the relevant directions of tensor flattening that may not be parallel either to rows or columns. When rearranging data so that flattening can be performed in the estimated directions, the signal subspace dimension is reduced, and the signal-to-noise ratio is improved. We adapt the bidimensional straight-line detection algorithm that estimates the HSI main directions, which are used to flatten the HSI tensor. We also generalize the quadtree partitioning to tensors in order to adapt the filtering to the image discontinuities. Comparative studies with MWF, wavelet thresholding, and channel-by-channel Wiener filtering show that our algorithm provides better performance while restoring impaired HYDICE HSIs.