一种加权增强的欠定盲辨识方法

针对欠定盲源分离混合矩阵问题,提出了一种基于二阶统计量平行因子分解,加权增强最小二乘法的欠定混合盲辨识方法。该算法不需要源信号满足稀疏性要求,仅在源信号满足相互独立和最多一个高斯信号的条件下,将独立源信号的空间协方差矩阵构建三阶张量,采用加权增强最小二乘法实现张量的标准分解,完成混合矩阵的估计。由于平行因子分解的唯一性在欠定条件下依然成立,该算法可以解决欠定盲源分离问题。仿真实验结果表明:提出的算法在计算欠定混合时具有很好的辨识效果,而且实现简单,可满足实际应用的要求。     

一种新型频域快速盲分离算法

为克服传统盲源分离算法分离效果差、计算量大且输出信号尺度模糊的缺点,提出了一种新型频域快速盲源分离算法.该算法在分析时域水声信号混合模型的基础上,构建新型频域混合模型,采用混合神经网络计算某一频率的分离矩阵,以此来估计全局分离矩阵.新算法较好地克服了尺度模糊问题,极大地减小了计算量,增加了分离算法的灵活性,分离性能较好.水声信号仿真实验和湖试实验均验证了算法的有效性.     

基于EEMD和ICA的单通道列车信号盲分离

针对列车混合故障的诊断,提出了一种基于集合平均经验分解（EEMD）和独立分量分析（ICA）的盲分离诊断方法。通过EEMD算法将混合信号分解为包含不同源信号特征的本征模态函数（IMF）,组成新的多维信号;用主成分分析准确估计源信号个数,解决了单通道信号盲分离的欠定问题;利用快速独立分量分析（Fast-ICA）算法实现了信号的盲分离。实验信号分别采用仿真信号和列车实验信号进行实验,实验结果表明,该算法可以有效地分离出列车的单故障信号。     

一种基于时域解相关的双通道语音分离算法

提出了一种基于时域解相关的卷积混合语音盲分离时域算法。该算法利用相关系数对语音信号进行时域去相关处理,同时基于二阶统计量完成双通道语音分离。该算法充分考虑了语音信号本身的相关性及不平稳性,收敛速度快,稳定性好,为减小计算量提出了该算法的递推改进算法。仿真实验表明,在对卷积混合双通道语音信号进行盲分离时,该算法是非常有效的。     

一种新的用于语音分离的盲源分离算法

提出了一种基于自然梯度的语音盲分离改进算法。该算法首先使用奇异值分解（SVD）的方法对观测信号进行预白化处理,而后使用自然梯度算法对预白化处理后的观测信号进行分离。通过计算机模拟试验,结果显示该算法能够有效地分离随机混合的自然语音信号。     

基于二阶统计和时间结构的盲信号分离方法

盲源分离(BSS)的目标就是在混合过程未知的情况下,仅仅依据观测得到的混合信号,恢复出不能直接观测的源信号。针对具有时间结构的源信号,即各个源信号分量满足空间上不相关但时间上相关,提出了一种基于二阶统计量的盲源分离方法。该方法首先对混合信号进行鲁棒预白化处理,其中依据最小描述长度准则对源信号的维数进行估计;然后通过对白化信号的时延协方差矩阵进行奇异值分解(SVD),从而实现源信号的盲分离。仿真中通过对一组语音信号的分离验证了算法的效果,并利用信号干扰比(SIR)和性能指标函数(PI)两个指标定量地对算法的性能进行了度量。     

基于FPGA模糊变步长的在线盲源分离算法的实现

针对变步长的在线盲分离的扩展Infomax算法,提出了模糊变步长在线盲分离的扩展Infomax算法,以MATLAB仿真为理论依据,即当信号逐渐被分离,峭度方差逐渐递减,而步长因子随峭度方差和峭度方差变化率的递减而平滑地减小。并以FPGA查找表来实现模糊变步长更新算法,用每一次FPGA内部串行的步长调整量来选通多路选择器的端口数据,以达到迭代步长更新的目的。通过仿真结果表明这种设计方法具有很强的实时性,可实时跟踪峭度方差曲线,从而能较好地实现超高斯和亚高新信号混合情况的盲分离,提高变步长的在线盲分离的收敛速度和精度。     

估计盲信源分离混合矩阵的通用方法

混合矩阵的估计是解决盲信源分离问题的关键一步,但现有研究中缺乏一种同时适用于适定、超定及欠定情况下混合矩阵估计的通用方法。根据张量标准分解的因子矩阵和盲信源分离混合矩阵的估计均存在幅值和排列顺序的不确定性这一性质,将混合矩阵的估计转化为观测信号统计量所组成张量的标准分解问题,标准分解采用循环最小化方法,通过交替最小二乘算法实现。理论分析和仿真实验表明,所提方法可有效解决适定、超定和欠定混合矩阵的估计,是一种估计盲信源分离混合矩阵的通用方法。     

基于核估计的超定混合共轭盲信号分离方法

当混合信号的个数多于源信号时,盲源分离模型中的混合矩阵被描述为一个超定矩阵,因此不能直接通过估计逆矩阵的方法来得到分离矩阵。针对该线性超定混合情况提出了一种基于共轭梯度的盲源分离方法。该方法基于最小互信息准则,通过对行满秩分离矩阵的奇异值分解而引入了超定盲源分离的代价函数。利用共轭梯度优化算法推导出了迭代计算分离矩阵的更新公式。在每次迭代计算中,利用随机变量概率密度估计的核函数法在线估计分离信号的评价函数。避免了诸多传统盲分离算法中只能凭经验选取特定的非线性函数来代替评价函数的问题。仿真结果验证了所提算法的有效性。     

基于时域去相关预处理的卷积混合语音盲分离算法

因语音信号自身的相关性及不平稳性,使卷积混合语音信号的盲分离变得十分困难.本文提出了一种基于时域去相关预处理的卷积混合语音盲分离时域算法.该算法采用自适应格型预测误差滤波器对语音信号进行时域去相关处理,同时基于空域去相关算法完成卷积混合语音分离.该算法充分考虑了语音信号自身的相关性及不平稳性,具有计算量小、收敛速度快、稳定性好的优点.仿真实验验证了该算法在对卷积混合语音信号进行盲分离时的有效性.     

基于回溯搜索优化的卷积混合语音盲分离

针对独立矢量分析（IVA）算法初始分离矩阵取值对分离性能影响较大的局限性,提出了基于回溯搜索优化的卷积混合语音盲分离算法。采用频域各频率点IVA分离信号的复数峭度和作为目标函数,利用回溯搜索优化算法（BSA）对初始分离矩阵进行优化调整,更好地实现了语音信号的盲分离。在分离过程中,采用复Givens旋转变换原理将对分离矩阵的求解转化为对旋转角度的求解,有效减少了BSA的参数编码维数,降低了优化求解难度。针对语音信号的卷积混合分离实验表明,该算法具有良好的分离效果,其分离性能较之基本IVA算法显著提升。     

一种新的联合块对角化卷积盲分离时域算法

提出一种基于高阶累积量联合块对角化的时域算法求解卷积混合盲信号分离问题。引入白化处理,将混叠矩阵转变成酉矩阵,混合信号转变为互不相关的,进而计算出其对应的一系列高阶累积量矩阵,通过最小化代价函数来实现高阶累积量矩阵联合块对角化的目的,在时域中解决超定卷积盲分离问题。实验表明,相比于经典的自然梯度算法,所提方法的分离精度更高,且运算速度也更快。     

Mixing Audiovisual Speech Processing and Blind Source Separation for the Extraction of Speech Signals From Convolutive Mixtures

Looking at the speaker's face can be useful to better hear a speech signal in noisy environment and extract it from competing sources before identification. This suggests that the visual signals of speech (movements of visible articulators) could be used in speech enhancement or extraction systems. In this paper, we present a novel algorithm plugging audiovisual coherence of speech signals, estimated by statistical tools, on audio blind source separation (BSS) techniques. This algorithm is applied to the difficult and realistic case of convolutive mixtures. The algorithm mainly works in the frequency (transform) domain, where the convolutive mixture becomes an additive mixture for each frequency channel. Frequency by frequency separation is made by an audio BSS algorithm. The audio and visual informations are modeled by a newly proposed statistical model. This model is then used to solve the standard source permutation and scale factor ambiguities encountered for each frequency after the audio blind separation stage. The proposed method is shown to be efficient in the case of 2 times 2 convolutive mixtures and offers promising perspectives for extracting a particular speech source of interest from complex mixtures     

基于稳健联合分块对角化的卷积盲分离

针对卷积盲分离问题,提出一种新的矩阵联合分块对角化(Joint block diagonalization, JBD)算法. 现有的迭代非正交联合分块对角化算法都存在不收敛的情况,本文利用分离矩阵的特殊结构确保其可逆性,使得算法的迭代过程稳定. 在已知矩阵分块结构的条件下,首先,将卷积盲分离模型写成瞬时形式,并说明其满足联合分块对角化结构; 然后,提出联合分块对角化的代价函数,依据代价函数的最小化等价于矩阵中每个分块的范数最小化, 将整个分离矩阵的迭代更新转化成每个分块的迭代更新;最后,利用最小化条件得到迭代算法. 实数和复数两种情况下的算法都进行了推导.基本实验验证了新算法在不同条件下的性能; 仿真实验中对在时域和频域都重叠的信号的卷积混合进行盲分离,实验结果验证了新算法具有更好的分离性能和更稳定的分离能力.     

Adaptive vectorial lifting concept for convolutive blind source separation

This paper describes a new multi-resolution approach for the blind separation of convolutive image mixtures in transform domain. The proposed method uses an Adaptive Vectorial case of Quincunx Lifting Scheme (AVQLS), based on wavelet decomposition, and a geometric unmixing algorithm. It proceeds in three steps: first, the mixed images are decomposed by AVQLS. Then, the unmixing algorithm is applied to the more relevant component to get a transformed estimate of the original images. An inverse transform is, thereafter, applied to obtain an estimate of the original images. Experiments carried out on medical images showed that the proposed method yields better separation results than many widely used blind source separation algorithms.     

Multichannel Nonnegative Matrix Factorization in Convolutive Mixtures for Audio Source Separation

We consider inference in a general data-driven object-based model of multichannel audio data, assumed generated as a possibly underdetermined convolutive mixture of source signals. We work in the short-time Fourier transform (STFT) domain, where convolution is routinely approximated as linear instantaneous mixing in each frequency band. Each source STFT is given a model inspired from nonnegative matrix factorization (NMF) with the Itakura–Saito divergence, which underlies a statistical model of superimposed Gaussian components. We address estimation of the mixing and source parameters using two methods. The first one consists of maximizing the exact joint likelihood of the multichannel data using an expectation-maximization (EM) algorithm. The second method consists of maximizing the sum of individual likelihoods of all channels using a multiplicative update algorithm inspired from NMF methodology. Our decomposition algorithms are applied to stereo audio source separation in various settings, covering blind and supervised separation, music and speech sources, synthetic instantaneous and convolutive mixtures, as well as professionally produced music recordings. Our EM method produces competitive results with respect to state-of-the-art as illustrated on two tasks from the international Signal Separation Evaluation Campaign (SiSEC 2008).      

Convolutive Speech Bases and Their Application to Supervised Speech Separation

In this paper, we present a convolutive basis decomposition method and its application on simultaneous speakers separation from monophonic recordings. The model we propose is a convolutive version of the nonnegative matrix factorization algorithm. Due to the nonnegativity constraint this type of coding is very well suited for intuitively and efficiently representing magnitude spectra. We present results that reveal the nature of these basis functions and we introduce their utility in separating monophonic mixtures of known speakers     

基于卷积非负矩阵部分联合分解的强噪声单声道语音分离

非负矩阵部分联合分解(Nonnegative matrix partial co-factorization, NMPCF)将指定源频谱作为边信息参与混合信号频谱的联合分解, 以帮助确定指定源的基向量进而提高信号分离性能.卷积非负矩阵分解(Convolutive nonnegative matrix factorization, CNMF)采用卷积基分解的方法进行矩阵分解, 在单声道语音分离方面取得较好的效果.为了实现强噪声条件下的语音分离, 本文结合以上两种算法的优势, 提出一种基于卷积非负矩阵部分联合分解(Convolutive nonnegative partial matrix co-factorization, CNMPCF)的单声道语音分离算法.本算法首先通过基音检测算法得到混合信号的语音起始点, 再据此确定混合信号中的纯噪声段, 最后将混合信号频谱和噪声频谱进行卷积非负矩阵部分联合分解, 得到语音基矩阵, 进而得到分离的语音频谱和时域信号.实验中, 混合语音信噪比(Signal noise ratio, SNR)选择以-3 dB为间隔从0 dB至-12 dB共5种SNR.实验结果表明, 在不同噪声类型和噪声强度条件下, 本文提出的CNMPCF方法相比于以上两种方法均有不同程度的提高.     

An Approach for Solving the Permutation Problem of Convolutive Blind Source Separation Based on Statistical Signal Models

In this paper, we present a new algorithm for solving the permutation ambiguity in convolutive blind source separation. Transformed to the frequency domain, existing algorithms can efficiently solve the reduction of the source separation problem into independent instantaneous separation in each frequency bin. However, this independency leads to the problem of correctly aligning these single bins. The new algorithm models the frequency-domain separated signals by means of the generalized Gaussian distribution and employs the small deviation of the parameters between neighboring bins for the detection of correct permutations. The performance of the algorithm will be demonstrated on synthetic and real-world data.