一个用于卷积混合模型的时频分析盲分离算法

提出一种基于时频分析的卷积混合盲分离算法.由于信号源与各传感器的距离不同,在传播的过程中会产生不同的幅度衰减和时间延迟.该算法用短时傅里叶变换对语音信号进行时频分析,将其中一个传感器信号作为参考信号,构造了源信号的幅度衰减向量和时间延迟向量.根据语音信号的时频域稀疏性,以这两个向量为特征,在时频域上对传感器信号进行聚类,再通过估计参考信号混合系数来获得源信号时频域表示,进一步得到源信号.该方法可以用于源信号数目大于传感器信号数目的情况.仿真实验证明,算法可以完成欠定情况下卷积混合信号的盲分离,分离结果令人满意.     

Underdetermined Convolutive Blind Source Separation via Time–Frequency Masking

In this paper, we consider the problem of separation of unknown number of sources from their underdetermined convolutive mixtures via time-frequency (TF) masking. We propose two algorithms, one for the estimation of the masks which are to be applied to the mixture in the TF domain for the separation of signals in the frequency domain, and the other for solving the permutation problem. The algorithm for mask estimation is based on the concept of angles in complex vector space. Unlike the previously reported methods, the algorithm does not require any estimation of the mixing matrix or the source positions for mask estimation. The algorithm clusters the mixture samples in the TF domain based on the Hermitian angle between the sample vector and a reference vector using the well known k -means or fuzzy c -means clustering algorithms. The membership functions so obtained from the clustering algorithms are directly used as the masks. The algorithm for solving the permutation problem clusters the estimated masks by using k-means clustering of small groups of nearby masks with overlap. The effectiveness of the algorithm in separating the sources, including collinear sources, from their underdetermined convolutive mixtures obtained in a real room environment, is demonstrated.     

Convolutive Blind Source Separation in the Frequency Domain Based on Sparse Representation

Convolutive blind source separation (CBSS) that exploits the sparsity of source signals in the frequency domain is addressed in this paper. We assume the sources follow complex Laplacian-like distribution for complex random variable, in which the real part and imaginary part of complex-valued source signals are not necessarily independent. Based on the maximum a posteriori (MAP) criterion, we propose a novel natural gradient method for complex sparse representation. Moreover, a new CBSS method is further developed based on complex sparse representation. The developed CBSS algorithm works in the frequency domain. Here, we assume that the source signals are sufficiently sparse in the frequency domain. If the sources are sufficiently sparse in the frequency domain and the filter length of mixing channels is relatively small and can be estimated, we can even achieve underdetermined CBSS. We illustrate the validity and performance of the proposed learning algorithm by several simulation examples.     

解卷积混合语音频域盲分离的次序问题新方法

多通道语音信号的混合往往是卷积混合,瞬时盲分离方法不能获得好的分离效果,而频域方法由于频率次序的问题使性能下降.本文采用时频掩模的方法得到各频点上具有确定次序的、但带有失真的分离信号,将其作为参考,与频域上解得的次序不定信号进行相关,从而获得精确的语音分离信号.实验表明:本文提出的方法能有效地解决频域盲分离的次序不确定性问题,得到精度更高的分离卷积混舍的语音信号.     

一种新的卷积混合信号的盲分离算法

研究了一种新的线性卷积混合信号的盲分离算法。该算法通过计算预白化观测数据的零时延和多时延自相关协方差矩阵,获得了多时延处理的二阶解相关统计信息。利用得到的二阶统计信息构建了两个对称正定矩阵,通过使用Cholesky分解和奇异值分解等一系列变换,得出了惟一存在的矩阵。理论分析表明,该矩阵可以使两个正定矩阵同时精确对角化。计算机仿真表明,该算法与已有算法相比,运算时间短,盲分离性能更优。     

基于贝叶斯网络理论的时变混合盲源分离算法

以状态空间模型作为信道的变化模型,研究了时变混合情况下非平稳信号的盲分离问题。首先将隐马尔可夫模型（HMM）和混合高斯（MOG）模型结合起来对具有动态结构和复杂分布的非平稳源信号进行建模,然后运用贝叶斯网络理论处理信道时变情况下独立成分分析（ICA）模型中各变量和参数之间的关系,提出了一种基于贝叶斯推断的可同时完成混合矩阵盲估计及源信号盲分离的算法,通过采用逼近方法有效地减小了算法计算量。计算机仿真试验证明本文算法的有效性。     

应用信号变化度的线性混合信号盲分离算法

论文首先给出了信号变化度的概念,并证明了信号变化度的一个性质:互相独立的一组源信号的线性混合信号的变化度介于源信号中的最小变化度和最大变化度之间。然后,利用矩阵广义特征值理论,给出了一种基于线性混合信号盲分离算法。该算法计算简单,具有闭解形式;并能分离源信号中既有亚高斯信号又有超高斯信号的情况。仿真结果表明该算法是有效的,并具有很好的分离性能。     

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     

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.     

基于修正离散傅里叶变换的频域卷积混合盲分离

针对频域卷积混合盲分离,依据所导出的卷积混合信号每帧的频域表示模型,提出了一种最小均方误差意义下的最优变换--修正离散傅里叶变换,用于代替频域卷积混合盲分离中常用的离散傅里叶变换.在每个频率片上,卷积混合信号的修正离散傅里叶变换系数在最小均方误差意义下最接近于源信号频谱的瞬时混合.相对于离散傅里叶变换系数,现有瞬时混合盲分离算法能从修正离散傅里叶变抉系数中更精确地估计各频率片上分离矩阵,从而提高现有频域卷积混合盲分离算法的分离性能.仿真结果证明了修正离散傅里叶变换对现有频域卷积混合盲分离算法的有效性.     

Spatio–Temporal FastICA Algorithms for the Blind Separation of Convolutive Mixtures

This paper derives two spatio-temporal extensions of the well-known FastICA algorithm of Hyvarinen and Oja that are applicable to the convolutive blind source separation task. Our time-domain algorithms combine multichannel spatio-temporal prewhitening via multistage least-squares linear prediction with novel adaptive procedures that impose paraunitary constraints on the multichannel separation filter. The techniques converge quickly to a separation solution without any step size selection or divergence difficulties, and unlike other methods, ours do not require special coefficient initialization procedures to obtain good separation performance. They also allow for the efficient reconstruction of individual signals as observed in the sensor measurements directly from the system parameters for single-input multiple-output blind source separation tasks. An analysis of one of the adaptive constraint procedures shows its fast convergence to a paraunitary filter bank solution. Numerical evaluations of the proposed algorithms and comparisons with several existing convolutive blind source separation techniques indicate the excellent relative performance of the proposed methods.     

Blind Separation of Underdetermined Convolutive Mixtures Using Their Time–Frequency Representation

This paper considers the blind separation of nonstationary sources in the underdetermined convolutive mixture case. We introduce, two methods based on the sparsity assumption of the sources in the time-frequency (TF) domain. The first one assumes that the sources are disjoint in the TF domain, i.e., there is at most one source signal present at a given point in the TF domain. In the second method, we relax this assumption by allowing the sources to be TF-nondisjoint to a certain extent. In particular, the number of sources present (active) at a TF point should be strictly less than the number of sensors. In that case, the separation can be achieved thanks to subspace projection which allows us to identify the active sources and to estimate their corresponding time-frequency distribution (TFD) values. Another contribution of this paper is a new estimation procedure for the mixing channel in the underdetermined case. Finally, numerical performance evaluations and comparisons of the proposed methods are provided highlighting their effectiveness.     

Acoustic Source Separation of Convolutive Mixtures Based on Intensity Vector Statistics

Various techniques have previously been proposed for the separation of convolutive mixtures. These techniques can be classified as stochastic, adaptive, and deterministic. Stochastic methods are computationally expensive since they require an iterative process for the calculation of the demixing filters based on a separation criterion that usually assumes that the source signals are statistically independent. Adaptive methods, such as the adaptive beamformers, also exploit signal properties in order to optimize a multichannel filter structure. However, these algorithms need initialization and time to converge. Deterministic methods, on the other hand, provide a closed-form solution based on the deterministic aspects of the problem, such as the channel characteristics and the source directions. This paper presents a technique that exploits the intensity vector statistics to achieve a nearly closed-form solution for the separation of the convolutive mixtures as recorded with a coincident microphone array. No assumptions are made on the signals, but it is assumed that the source directions are known a priori. Directivity functions based on von Mises functions are designed for beamforming depending on the circular statistics of the calculated intensity vectors. Numerical evaluation results were presented for various speech and instrument sounds and source positions in two reverberant rooms.     

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).      

欠定条件下的盲分离算法

盲信号分离中当源信号个数大于观测信号个数,且源信号不是足够稀疏时,如果利用聚类算法进行分离,分离效果将会变差。为此提出一种在此欠定条件下新的盲信号分离算法。利用源信号的“稀疏性”估计混合矩阵,然后简化混合矩阵构造新的混合模型。由于源信号间具有的独立性,使得可以在新的混合模型中从观察信号的自相关函数中估计出源信号的频谱,从而达到分离出源信号的目的,且分离效果优于聚类算法。最后给出仿真试验实例,试验结果验证了算法的有效性。     

JADE盲信号分离算法失效问题研究

分离（或解混合）矩阵的学习算法是盲信号分离的关键技术,矩阵联合对角化的预白化JADE算法是一种基于四阶累计量的学习算法.本文在分析该算法原理的基础上,从理论上找出了算法失效的原因,即源信号相关性越强,JADE盲信号分离算法失效问题越严重,并通过仿真实验证明了理论分析结果的正确性.     

几种盲源分离算法的比较

文章研究了基于高阶统计量的FastlCA算法、基于信息理论的Infomax算法和基于四阶统计量的JADE算法等几种典型盲源分离算法在噪声环境下的分离性能.比较结果显示各种算法分离噪声信号的性能是不同的.对于某些混有一定噪声的特定信号,某些算法的分离性能要优于其他信号.研究结果还表明,基于信噪比和信号的特性来选择盲源分离算法能够给出令人满意的分离效果.     

Regularization Methods for Blind Deconvolution and Blind Source Separation Problems

This paper is devoted to blind deconvolution and blind separation problems. Blind deconvolution is the identification of a point spread function and an input signal from an observation of their convolution. Blind source separation is the recovery of a vector of input signals from a vector of observed signals, which are mixed by a linear (unknown) operator. We show that both problems are paradigms of nonlinear ill-posed problems. Consequently, regularization techniques have to be used for stable numerical reconstructions. In this paper we develop a rigorous convergence analysis for regularization techniques for the solution of blind deconvolution and blind separation problems. Convergence of regularized point spread functions and signals to a solution is established and a convergence rate result in dependence of the noise level is presented. Moreover, we prove convergence of the alternating minimization algorithm for the numerical solution of regularized blind deconvolution problems and present some numerical examples. Moreover, we show that many neural network approaches for blind inversion can be considered in the framework of regularization theory. Date received: August 17, 1999. Date revised: September 1, 2000.     

Blind Source Separation Exploiting Higher-Order Frequency Dependencies

Blind source separation (BSS) is a challenging problem in real-world environments where sources are time delayed and convolved. The problem becomes more difficult in very reverberant conditions, with an increasing number of sources, and geometric configurations of the sources such that finding directionality is not sufficient for source separation. In this paper, we propose a new algorithm that exploits higher order frequency dependencies of source signals in order to separate them when they are mixed. In the frequency domain, this formulation assumes that dependencies exist between frequency bins instead of defining independence for each frequency bin. In this manner, we can avoid the well-known frequency permutation problem. To derive the learning algorithm, we define a cost function, which is an extension of mutual information between multivariate random variables. By introducing a source prior that models the inherent frequency dependencies, we obtain a simple form of a multivariate score function. In experiments, we generate simulated data with various kinds of sources in various environments. We evaluate the performances and compare it with other well-known algorithms. The results show the proposed algorithm outperforms the others in most cases. The algorithm is also able to accurately recover six sources with six microphones. In this case, we can obtain about 16-dB signal-to-interference ratio (SIR) improvement. Similar performance is observed in real conference room recordings with three human speakers reading sentences and one loudspeaker playing music