A blind source separation algorithm based on negentropy and signal noise ratio

A novel blind source separation （BSS） algorithm based on the combination of negentropy and signal noise ratio （SNR） is presented to solve the deficiency of the traditional independent component analysis （ICA） algorithm after the introduction of the principle and algorithm of ICA. The main formulas in the novel algorithm are elaborated and the idiographic steps of the algorithm are given. Then the computer simulation is used to test the performance of this algorithm. Both the traditional FastlCA algorithm and the novel ICA algorithm are applied to separate mixed signal data. Experiment results show the novel method has a better performance in separating signals than the traditional FastlCA algorithm based on negentropy. The novel algorithm could estimate the source signals from the mixed signals more precisely.

A Modal Identification Algorithm Combining Blind Source Separation and State Space Realization

A modal identification algorithm is developed, combining techniques from Second Order Blind Source Separation (SOBSS) and State Space Realization (SSR) theory. In this hybrid algorithm, a set of correlation matrices is generated using time-shifted, analytic data and assembled into several Hankel matrices. Dissimilar left and right matrices are found, which diagonalize the set of nonhermetian Hankel matrices. The complex-valued modal matrix is obtained from this decomposition. The modal responses, modal auto-correlation functions and discrete-time plant matrix (in state space modal form) are subsequently identified. System eigenvalues are computed from the plant matrix to obtain the natural frequencies and modal fractions of critical damping. Joint Approximate Diagonalization (JAD) of the Hankel matrices enables the under determined (more modes than sensors) problem to be effectively treated without restrictions on the number of sensors required. Because the analytic signal is used, the redundant complex conjugate pairs are eliminated, reducing the system order (number of modes) to be identified half. This enables smaller Hankel matrix sizes and reduced computational effort. The modal auto-correlation functions provide an expedient means of screening out spurious computational modes or modes corresponding to noise sources, eliminating the need for a consistency diagram. In addition, the reduction in the number of modes enables the modal responses to be identified when there are at least as many sensors as independent (not including conjugate pairs) modes. A further benefit of the algorithm is that identification of dissimilar left and right diagonalizers preclude the need for windowing of the analytic data. The effectiveness of the new modal identification method is demonstrated using vibration data from a 6 DOF simulation, 4-story building simulation and the Heritage court tower building.

For LEO Satellite Networks: Intelligent Interference Sensing and Signal Reconstruction Based on Blind Separation Technology

In LEO satellite communication networks,the number of satellites has increased sharply, the relative velocity of satellites is very fast, then electronic signal aliasing occurs from time to time. Those aliasing signals make the receiving ability of the signal receiver worse, the signal processing ability weaker,and the anti-interference ability of the communication system lower. Aiming at the above problems, to save communication resources and improve communication efficiency, and considering the irregularity of interference signals, the underdetermined blind separation technology can effectively deal with the problem of interference sensing and signal reconstruction in this scenario. In order to improve the stability of source signal separation and the security of information transmission, a greedy optimization algorithm can be executed. At the same time, to improve network information transmission efficiency and prevent algorithms from getting trapped in local optima, delete low-energy points during each iteration process. Ultimately, simulation experiments validate that the algorithm presented in this paper enhances both the transmission efficiency of the network transmission system and the security of the communication system, achieving the process of interference sensing and signal reconstruction in the LEO satellite communication system.

Blind source separation algorithm for communication complex signals in communication reconnaissance

Most blind source separation algorithms are only applicable to real signals,while in communication reconnaissance processed signals are complex.To solve this problem,a blind source separation algorithm for communication complex signals is deduced,which is obtained by adopting the Kullback-Leibler divergence to measure the signals’independence.On the other hand,the performance of natural gradient is better than that of stochastic gradient,thus the natural gradient of the cost function is used to optimize the algorithm.According to the conclusion that the signal’s mixing matrix after whitening is orthogonal,we deduce the iterative algorithm by constraining the separating matrix to an orthogonal matrix.Simulation results show that this algorithm can efficiently separate the source signals even in noise circumstances.

A robust clustering algorithm for underdetermined blind separation of sparse sources

In underdetermined blind source separation, more sources are to be estimated from less observed mixtures without knowing source signals and the mixing matrix. This paper presents a robust clustering algorithm for underdetermined blind separation of sparse sources with unknown number of sources in the presence of noise. It uses the robust competitive agglomeration （RCA） algorithm to estimate the source number and the mixing matrix, and the source signals then are recovered by using the interior point linear programming. Simulation results show good performance of the proposed algorithm for underdetermined blind sources separation （UBSS）.

BLIND SPEECH SEPARATION FOR ROBOTS WITH INTELLIGENT HUMAN-MACHINE INTERACTION

Speech recognition rate will deteriorate greatly in human-machine interaction when the speaker's speech mixes with a bystander's voice. This paper proposes a time-frequency approach for Blind Source Seperation (BSS) for intelligent Human-Machine Interaction(HMI). Main idea of the algorithm is to simultaneously diagonalize the correlation matrix of the pre-whitened signals at different time delays for every frequency bins in time-frequency domain. The prososed method has two merits: (1) fast convergence speed; (2) high signal to interference ratio of the separated signals. Numerical evaluations are used to compare the performance of the proposed algorithm with two other deconvolution algorithms. An efficient algorithm to resolve permutation ambiguity is also proposed in this paper. The algorithm proposed saves more than 10% of computational time with properly selected parameters and achieves good performances for both simulated convolutive mixtures and real room recorded speeches.

A TIME-FREQUENCY BLIND SEPARATION METHOD FOR UNDERDETERMINED SPEECH MIXTURES

The proposed Blind Source Separation method(BSS),based on sparse representations,fuses time-frequency analysis and the clustering approach to separate underdetermined speech mixtures in the anechoic case regardless of the number of sources.The method remedies the insufficiency of the Degenerate Unmixing Estimation Technique(DUET) which assumes the number of sources a priori.In the proposed algorithm,the Short-Time Fourier Transform(STFT) is used to obtain the sparse rep-resentations,a clustering method called Unsupervised Robust C-Prototypes(URCP) which can ac-curately identify multiple clusters regardless of the number of them is adopted to replace the histo-gram-based technique in DUET,and the binary time-frequency masks are constructed to separate the mixtures.Experimental results indicate that the proposed method results in a substantial increase in the average Signal-to-Interference Ratio(SIR),and maintains good speech quality in the separation results.

Initialization for NMF-Based Audio Source Separation Using Priors on Encoding Vectors

Nonnegative matrix factorization(NMF)has shown good performances on blind audio source separation(BASS).While the NMF analysis is a non-convex optimization problem when both the basis and encoding matrices need to be estimated simultaneously,the source separation step of the NMF-based BASS with a fixed basis matrix has been considered convex.However,because the basis matrix for the BASS is typically constructed by concatenating the basis matrices trained with individual source signals,the subspace spanned by the basis vectors for one source may overlap with that for other sources.In this paper,we have shown that the resulting encoding vector is not unique when the subspaces spanned by basis vectors for the sources overlap,which implies that the initialization of the encoding vector in the source separation stage is not trivial.Furthermore,we propose a novel method to initialize the encoding vector for the separation step based on the prior model of the encoding vector.Experimental results showed that the proposed method outperformed the uniform random initialization by 1.09 and 2.21dB in the source-to-distortion ratio,and 0.20 and 0.23 in PESQ scores for supervised and semi-supervised cases,respectively.

Underdetermined Blind Mixing Matrix Estimation Using STWP Analysis for Speech Source Signals

Wavelet packets decompose signals in to broader components using linear spectral bisecting. Mixing matrix is the key issue in the Blind Source Separation (BSS) literature especially in under-determined cases. In this paper, we propose a simple and novel method in Short Time Wavelet Packet (STWP) analysis to estimate blindly the mixing matrix of speech signals from noise free linear mixtures in over-complete cases. In this paper, the Laplacian model is considered in short time-wavelet packets and is applied to each histogram of packets. Expectation Maximization (EM) algorithm is used to train the model and calculate the model parameters. In our simulations, comparison with the other recent results will be computed and it is shown that our results are better than others. It is shown that complexity of computation of model is decreased and consequently the speed of convergence is increased.

Improved statistical sparse decomposition principle method for underdetermined blind source signal recovery

Aiming at the statistical sparse decomposition principle(SSDP) method for underdetermined blind source signal recovery with problem of requiring the number of active signals equal to that of the observed signals, which leading to the application bound of SSDP is very finite, an improved SSDP(ISSDP) method is proposed. Based on the principle of recovering the source signals by minimizing the correlation coefficients within a fixed time interval, the selection method of mixing matrix’s column vectors used for signal recovery is modified, which enables the choose of mixing matrix’s column vectors according to the number of active source signals self-adaptively. By simulation experiments, the proposed method is validated. The proposed method is applicable to the case where the number of active signals is equal to or less than that of observed signals, which is a new way for underdetermined blind source signal recovery.

SPEECH ENHANCEMENT BASED ON SECOND ORDER ARCHITECTURE AND INFORMATION MAXIMIZATION THEORY

Based on the idea of adaptive noise cancellation (ANC), a second order architecture is proposed for speech enhancement. According as the Information Maximization theory, the corresponding gradient descend algorithm is proposed. With real speech signals in the simulation, the new algorithm demonstrates its good performance in speech enhancement. The main advantage of the new architecture is that clean speech signals can be got with less distortion.

基于稀疏编码的复杂机械振动信号盲分离方法

复杂机械振动信号激励源较多,故源信号之间互为相关源,且较难满足统计独立特性,导致传统盲源分离方法分离效果不佳。对此,提出一种基于信号稀疏编码的机械振动信号盲分离方法。盲源分离的关键在于对混合矩阵的精确估计,然而机械振源中相关成分的存在严重影响混合矩阵的估计。对此,首先对观测信号进行短时傅里叶变换,增加信号稀疏性;然后利用稀疏编码筛选出具备直线聚类特性的时频观测点,利用K均值(K-means)聚类法找到聚类中心;最后利用所提筛选规则找到估计的混合矩阵,重构出源信号。通过对往复压缩机故障数据的分析,验证了所提方法有效性。

动态变化混叠模型下盲源分离中的源数估计

在进行欠定盲分离时,特别是对于源信号数目及混合矩阵动态变化的情况,常规的欠定盲分离及源数估计方法不能对源信号数目的变化时刻做出判断,因此很难实现动态变化的源信号数目实时和准确的估计。针对这个问题,提出了一种动态变化混叠模型下欠定盲源分离中的源数估计方法。首先,建立动态变化混叠情形下盲源分离的数学模型及动态标识矩阵。其次,基于构建的动态标识矩阵统计和判断动态源信号数目的变化情况。最后,通过分段时间内多维观测矢量采样点聚类区间局部峰值统计,实现动态变化混叠模型下盲源分离中的源信号数目的有效估计。仿真结果表明,该方法能有效实现动态变化混叠模型下欠定盲源分离中的源数估计,并且信号估计效果良好。

基于参数估计和Kalman滤波的单通道盲源分离算法

针对存在频谱混叠通信信号的单通道盲源分离(single channel blind source separation,SCBSS)问题,提出一种基于参数估计和Kalman滤波的SCBSS算法。首先,针对根多重信号分类(root multiple signal classification,Root-MUSIC)算法在相近载频估计方面的局限性,提出一种自适应的Root-MUSIC算法,对接收到的盲混合信号的源信号数目和载频进行估计;其次,将Kalman滤波的思想引入到SCBSS算法中,根据估计得到的源信号参数构造信号模型,将其作为Kalman滤波系统的观测向量,执行“时间更新”和“测量更新”两个过程,得到源信号的最佳估计,实现单通道盲源分离。仿真结果表明,所提算法能够有效地从存在频谱混叠的单路接收信号中准确地分离出多路源信号,比传统的算法分离精度高,运算速度快。

基于两步单源点筛选的改进退化解混和估计算法

退化解混和估计(Degenerate unmixing estimation technique,DUET)算法是一种典型的欠定盲源分离算法,其采用的二进制时频掩蔽会保留部分干扰信号。提出了基于两步单源点筛选的改进DUET算法,首先使用余弦角算法进行单源点筛选,再采用计算相似度的方法进行第二步单源点筛选。通过两步单源点筛选获得更精确的目标信号和干扰信号后,设计用于抵消干扰信号的滤波器取代DUET中的二进制时频掩蔽,达到抑制干扰信号和提取目标信号的目的。仿真实验结果表明,该方法在正定盲源分离和欠定盲源分离两种情况下都有较优的盲源分离性能。

基于JADE-斜投影的鲁棒波束形成算法

针对矩阵重构类波束形成算法对阵列幅相误差敏感的问题,提出一种基于盲源信号分离和斜投影的矩阵重构鲁棒波束形成算法。首先,依靠盲源分离技术得到接收信号和混合矩阵,结合期望信号先验信息完成混合矩阵中信号导向矢量的搜索。然后,利用盲源分离得到的信号协方差矩阵完成阵列幅相误差估计。最后,基于幅相误差校准的混合矩阵和斜投影思想,构建各干扰的斜投影算子,将接收数据分别向干扰斜投影空间进行投影,得到对应的干扰信号,完成干扰噪声协方差矩阵重构。仿真结果表明,所提方法对阵列幅相误差具有较好的鲁棒性,验证了算法的有效性。

基于QR分解的类Jacobi联合对角化算法

为提高实矩阵集的近似联合对角化的盲源分离性能,避免平凡解,提出了一种基于QR分解的类Jacobi联合对角化算法.利用QR分解的数值稳定性,采用Jacobi旋转矩阵,将分离矩阵分解为多个初等三角矩阵和正交矩阵的乘积,利用Jacobi旋转矩阵的结构及矩阵变换后的相关元素求解最优参数,将高维矩阵最小化问题转化为一系列低维矩阵子问题,提升源信号恢复精度.通过求解简化的Frobenius范数目标函数降低算法复杂度.混合心电信号仿真结果表明,与QRJ2D,LUCJD,EGJLUD算法相比,本文算法在分离精度和收敛速度方面均有一定优势.

FCM-FastICA的点蚀声发射信号分离识别方法

金属点蚀是一种破坏性和隐患较大的设备损伤形式。点蚀会产生声发射信号。点蚀过程中产生的多种声源类型会造成信号混叠,影响腐蚀进程的判断。针对点蚀信号混叠问题,提出一种模糊C均值聚类与快速独立分量分析算法相结合的点蚀信号分离识别方法。通过分析单、双点蚀声发射数据将点蚀分为钝化膜破裂阶段、点蚀诱导成核及发展阶段,由聚类确定信号类别并用快速独立分量分析分离混合信号,利用相关性函数验证分离效果。结果表明:单点蚀过程存在3类原信号,双点蚀过程存在7类信号,其中包含单个信号与混合信号;单个信号与原信号相关性极高,达到0.8以上,混合信号的分离分量与原信号相关性达到0.6以上,分离效果较好。该方法可对点蚀混合信号进行有效分离和识别,为腐蚀进程判断提供支持。

基于自适应参数优化RSSD-CYCBD的行星齿轮箱复合故障诊断

针对行星齿轮箱多振源耦合导致故障源辨识困难、较弱故障特征容易被噪声和较强故障特征掩盖,以及由传播路径引起的信号衰减导致的故障特征微弱等问题,提出一种自适应参数优化的共振稀疏分解(RSSD)和最大二阶循环平稳盲解卷积(CYCBD)的行星齿轮箱多故障耦合信号分离及诊断算法。根据轴承和齿轮故障的不同共振属性,用RSSD算法将多故障耦合信号分解为包含齿轮故障特征的高共振分量和主要包含轴承故障特征的低共振分量后,通过CYCBD算法分别对高、低分量进行解卷积,消除传播路径影响和噪声干扰,实现微弱故障特征的增强和提取。特别地,针对RSSD和CYCBD中参数优化困难、依赖人工经验和自适应差等问题,使用基于松鼠算法(SSA)对参数进行自适应优化选取,设计了融合包络谱峭度、自相关函数最大值均方根和特征频率比在内的复合指标作为优化目标。对解卷积后的信号进行包络解调提取故障特征频率,识别不同故障源。通过行星齿轮箱多故障模拟信号和实测信号验证了所提算法的有效性和可行性,进一步地,将所提算法集成在边缘计算设备中,为行星齿轮箱等旋转机械的状态检测诊断及远程运维提供解决方案。

基于欠定盲源分离的双路音频信号噪声自适应分离

当多个源信号同时存在于同一频段或时间域内时,它们可能会相互干扰,导致信号混叠。这种情况下使用双路音频传感器进行捕捉,无法准确地捕捉到所有源信号的信息,导致分离过程具有不确定性。对此,提出一种基于欠定盲源分离的双路音频信号噪声自适应分离方法。首先,构建欠定盲源分离模型,基于小波包变换分解和重构信号获取信号分量,并依据信号和分量之间的互相关系数筛选分解后的分量,删除其中的冗余分量后生成新的观测信号;然后,依据贝叶斯信息准则的奇异值分解方法估计该源信号的数量,将其转换为正定白化信号;最后,利用快速独立成分分析法将该信号分类,实现双路音频信号噪声自适应分离。测试结果显示:所提方法能够在保证信号质量的前提下完成信号变换处理,信干比均在15 dB以上;筛选后保留的各个分量相关系数均在0.65以上,有效地完成了对信号噪声的分离。