基于Givens矩阵和联合非线性不相关的盲源分离新算法

针对源信号统计独立的盲源分离(Blind Source Separation,BSS)问题,提出了一种基于Givens矩阵和联合非线性不相关的盲源分离新算法.由于分离信号独立性的度量是影响算法有效性的重要因素,因此首先提出了一种改进的度量独立性的方法,该方法以独立源信号的联合非线性不相关来度量独立性;其次,结合Givens矩阵可以对分离矩阵施加正交性约束且能减少要估计参数个数的性质,将盲源分离问题转化成无约束优化问题,并利用拟牛顿法中的BFGS算法求解该无约束优化问题,得到分离矩阵;最后,通过模拟混合信号和真实语音混合信号的分离实验验证了该算法的有效性.     

语音增强的计算机仿真

研究关于盲源分离的特征向量分离算法在语音增强的应用,传统的方法对混合的语音信号很难进行有效的分离,而在实际中很多场合都需要对语音信号进行增强.为消除噪音,提高清晰度,使用的盲源分离算法却正能实现传统方法难以实现的技术.运用一种盲源分离的特征向量分离算法来进行语音增强,并且对实际的两个语音信号运用该算法进行了混合语音信号的分离增强实验,利用MATLLAB软件对混合语音信号进行了盲源分离的特征向量分离算法的仿真,可从混合语音信号分离出了两个原始语音信号.证明了盲源分离算法应用于语音分离的可行性,为盲源分离应用于语音增强提供了参考依据.     

PID控制原理在语音信号盲源分离中的应用

语音信号在非平稳系统中是动态混合的,为了实时抑制盲源分离过程中的非平稳混合扰动,加快收敛速度,减小稳态误差,提出了一种应用PID控制原理的自适应盲源分离算法。依据一种无预处理的自适应盲源分离算法建立PID控制模型,调节学习速率,跟踪语音信号的分离过程,实时减小由非平稳混合引入的分离误差,动态更新分离矩阵。在混合矩阵缓变和突变两种情形下分别对PID参数整定和语音信号的分离进行仿真分析,结合经典算法对比提出算法的性能。仿真与对比结果表明,提出的算法适用于非平稳混合系统语音信号的分离,算法性能较经典算法有改善。     

基于最大信噪比的盲源分离算法

提出一种新的低计算复杂度的瞬时线性混叠信号的盲分离算法,该算法利用统计独立信号完全分离时信噪比量大作为分离准则。源信号用估计信号的滑动平均代替,把源信号和噪声信号协方差矩阵的函数表示成广义特征值问题,通过广义特征值问题求解分离矩阵不需要任何迭代运算。和典型的信息理论方法相比,该算法的优点是具有非常低的计算复杂度。计算机模拟实验证明,该算法能够分离线性混合的超高斯和亚高斯源信号,并且可以有效地分离语音信号。     

基于Givens变换矩阵的时间结构信号盲源分离新算法

对于时间结构信号的盲源分离(Blind Source Separation,BSS),独立成分分析(Independent Component Analysis,ICA)是十分有效的方法。在对观测信号白化处理后,ICA的关键是寻找去除高阶相关性的正交分离矩阵。鉴于任意维数正交矩阵可以表示为Givens变换矩阵的乘积,提出了一种新的时间结构信号盲源分离算法。首先,利用Givens变换矩阵参数化表示正交分离矩阵,减少了要估计参数的个数;其次,以多步时延协方差矩阵的联合近似对角化为目标函数,将盲源分离问题转化为无约束优化问题,并利用拟牛顿法中的BFGS算法对Givens变换矩阵中的参数进行估计,得到分离矩阵;最后,以实际的混合语音信号分离做仿真实验,验证了该算法对时间结构信号的盲源分离是有效的。     

弱稀疏性下的欠定语音盲分离方法

针对语音信号的弱稀疏性,提出一种新的基于混合矩阵估计的欠定语音盲分离方法。该方法通过主成分分析检测只有一个源信号存在时的时频点并用于估计混合矩阵,从而克服语音信号稀疏性变弱时的影响,提高混合矩阵估计精度。结合子空间法重构源信号,进一步提高分离性能,并从几何角度证明子空间方法,仿真结果表明该方法的分离性能优于Cluster-UBSS,且鲁棒性更好。     

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.     

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。     

基于准正交原理的多信源少观测源的盲语音信号分离

信号源个数多于观测信号个数情况下的盲源分离问题是盲信号分离中的一个难题,也是一个很实际的问题。论文在A.Hyvrinen提出的一种基于准正交原理的盲分离算法基础上,指出当混合矩阵的基矢量不满足准正交性时,可以对观测信号预白化,使混合矩阵的基矢量的准正交性得以很大提高。然后将此方法用于多信源少观测源情况下的混合语音信号分离。实验分为两个过程:(1)估计混合矩阵;(2)用最大后验概率的估计方法分离源语音信号。实验结果证明了该算法能够有效用于高维情况下多信源少观测源的盲语音信号分离。     

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。     

基于子带无迹粒子滤波的语音增强算法

结合多采样率系统理论中的子带分解技术与贝叶斯估计理论中的无迹粒子滤波技术,提出了一种基于子带无迹粒子滤波的语音增强方法。该方法首先将语音信号分解成子带信号,建立各子带信号的低阶时变自回归模型;然后利用无迹粒子滤波估计模型参数,对子带信号进行滤波处理;最后根据滤波后的子带信号重构语音信号,实现语音增强。仿真结果表明,该方法能明显改善语音信号的信噪比和质量,且易于实现。     

Improved subband-forward algorithm for acoustic noise reduction and speech quality enhancement

This paper addresses the problem of speech enhancement and acoustic noise reduction by adaptive filtering algorithms. Recently, we have proposed a new Forward blind source separation algorithm that enhances very noisy speech signals with a subband approach. In this paper, we propose a new variable subband step-sizes algorithm that allows improving the previous algorithm behaviour when the number of subband is selected high. This new proposed algorithm is based on recursive formulas to compute the new variable step-sizes of the cross-coupling filters by using the decorrelation criterion between the estimated sub-signals at each subband output. This new algorithm has shown an important improvement in the steady state and the mean square error values. Along this paper, we present the obtained simulation results by the proposed algorithm that confirm its superiority in comparison with its original version that employs fixed step-sizes of the cross-coupling adaptive filters and with another fullband algorithm.     

混响环境下的宽带波束形成语音增强方法

结合小波滤波器组理论和自适应波束形成技术,提出了一种基于宽带波束形成的麦克风阵列语音增强方法。该方法利用小波分析滤波器组将含噪语音信号变换到小波域;进行小波域阵列自适应波束形成;通过小波综合滤波器组重构增强后的语音信号。计算机仿真实验验证了该方法的有效性。     

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.     

Nonlinear blind source separation using higher order statistics anda genetic algorithm

This paper presents a novel method for blindly separating unobservable independent source signals from their nonlinear mixtures. The demixing system is modeled using a parameterized neural network whose parameters can be determined under the criterion of independence of its outputs. Two cost functions based on higher order statistics are established to measure the statistical dependence of the outputs of the demixing system. The proposed method utilizes a genetic algorithm (GA) to minimize the highly nonlinear and nonconvex cost functions. The GA-based global optimization technique is able to obtain superior separation solutions to the nonlinear blind separation problem from any random initial values. Compared to conventional gradient-based approaches, the GA-based approach for blind source separation is characterized by high accuracy, robustness, and convergence rate. In particular, it is very suitable for the case of limited available data. Simulation results are discussed to demonstrate that the proposed GA-based approach is capable of separating independent sources from their nonlinear mixtures generated by a parametric separation model     

Blind Subband Beamforming With Time-Delay Constraints for Moving Source Speech Enhancement

A new robust microphone array method to enhance speech signals generated by a moving person in a noisy environment is presented. This blind approach is based on a two-stage scheme. First, a subband time-delay estimation method is used to localize the dominant speech source. The second stage involves speech enhancement, based on the acquired spatial information, by means of a soft-constrained subband beamformer. The novelty of the proposed method involves considering the spatial spreading of the sound source as equivalent to a time-delay spreading, thus, allowing for the estimated intersensor time-delays to be directly used in the beamforming operations. In comparison to previous approaches, this new method requires no special array geometry, knowledge of the array manifold, or acquisition of calibration data to adapt the array weights. Furthermore, such a scheme allows for the beamformer to efficiently adapt to speaker movement. The robustness of the time-delay estimation of speech signals in high noise levels is improved by making use of the non-Gaussian nature of speech trough a subband Kurtosis-weighted structure. Evaluation in a real environment with a moving speaker shows promising results, with suppression levels of up to 16 dB for background noise and interfering (speech) signals, associated to a relatively small effect of speech distortion.     

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.     

A perceptually motivated stationary wavelet packet filterbank using improved spectral over-subtraction for enhancement of speech in various noise environments

In this paper, we propose a speech enhancement method where the front-end decomposition of the input speech is performed by temporally processing using a filterbank. The proposed method incorporates a perceptually motivated stationary wavelet packet filterbank (PM-SWPFB) and an improved spectral over-subtraction (I-SOS) algorithm for the enhancement of speech in various noise environments. The stationary wavelet packet transform (SWPT) is a shift invariant transform. The PM-SWPFB is obtained by selecting the stationary wavelet packet tree in such a manner that it matches closely the non-linear resolution of the critical band structure of the psychoacoustic model. After the decomposition of the input speech, the I-SOS algorithm is applied in each subband, separately for the estimation of speech. The I-SOS uses a continuous noise estimation approach and estimate noise power from each subband without the need of explicit speech silence detection. The subband noise power is estimated and updated by adaptively smoothing the noisy signal power. The smoothing parameter in each subband is controlled by a function of the estimated signal-to-noise ratio (SNR). The performance of the proposed speech enhancement method is tested on speech signals degraded by various real-world noises. Using objective speech quality measures (SNR, segmental SNR (SegSNR), perceptual evaluation of speech quality (PESQ) score), and spectrograms with informal listening tests, we show that the proposed speech enhancement method outperforms than the spectral subtractive-type algorithms and improves quality and intelligibility of the enhanced speech.     

基于子带广义旁瓣相消器的麦克风阵列语音增强*

为了加快基于广义旁瓣相消器的麦克风阵列语音增强系统的收敛速度,将其自适应模块的输入信号分解到子带以进行处理,并将多通道维纳滤波器引入广义旁瓣相消器的非自适应支路,以更有效地抑制非相干噪声。实际测试结果表明,相对于基于全带广义旁瓣相消器的麦克风阵列语音增强系统,采用该子带广义旁瓣相消器结构的语音增强系统具有更快的收敛速度和更高的输出信噪比。