面向卷积混叠环境下的盲源分离新方法

卷积混叠环境下的盲源分离(Blind source separation, BSS)是一个极具挑战性和实际意义的问题. 本文在独立分量分析框架下, 建立非负矩阵分解(Nonnegative matrix factorization, NMF)模型, 设计新的优化目标函数, 通过严格的数学理论推导, 得到新的模型参数更新规则; 并对解混叠矩阵进行标准化处理, 避免幅度歧义性问题; 在源信号的重构阶段, 通过实时更新非负矩阵分解模型参数, 避免源信号的排序歧义性问题. 实验结果验证了所提算法在分离中英文语音混叠信号、音乐混叠信号时的有效性和优越性.     

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.     

卷积混合盲源分离中1种改进的排序模糊消除算法

频域方法可以有效地解决卷积混合盲源分离问题.针对频域方法中存在排序模糊,基于分离信号相邻频点功率谱密度的相关性较高的原理,提出1种改进的排序模糊消除算法.相比于原算法,扩展了参考频点的取值范围,同时还采用了1种置信度量方法,能够获得更准确的排序估计.仿真实验表明所提算法有效地消除了排序模糊,并且能够纠正某一频点排序的突...     

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.     

基于新型混合模型的欠定盲分离方法

针对欠定盲分离问题,提出了一种新的源恢复方法. 在时频域局部区域采用复高斯分布对源信号进行建模,将语音信号的稀疏性和局部平稳性结合在一起,提出了一种新的混合模型来描述观测信号在局部区域的概率分布.通过该模型,将每个时频点的源信号状态的判断问题转换成模型的参数估计和后验概率的计算问题,最后通过子混合矩阵的逆恢复出源信号. 实验结果表明,该方法具有很快的收敛速度,并且比已有方法具有更好的分离性能.     

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.     

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.     

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.     

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。