New automatic forward and backward blind sources separation algorithms for noise reduction and speech enhancement

In this paper, we consider the speech enhancement problem in a moving car through a blind source separation (BSS) scheme involving two spaced microphones. The forward and backward blind source separation structures often use manual voice activity detector (MVAD) systems to control the adaptation of the separating adaptive filters. In this paper, we propose two new automatic voice activity detector (AVAD) systems that allow adapting the original forward and backward BSS structures automatically. The proposed AVAD systems are based on the use of the forward BSS structure to estimate the optimal values of the separating adaptive filters step-sizes. Moreover, the new proposed algorithms are stable and could be used even in very noisy conditions. Intensive experiments are carried out with these two new proposed algorithms to validate their good performances in speech enhancement and noise reduction applications. The presented experiments are based on the system mismatch, the cepstral distance and the output signal-to-noise ratio criteria evaluations. The obtained results show the good performances of the proposed algorithms in comparison with their original versions, where manual VAD systems are used.     

一种基于EEMD域统计模型的话音激活检测算法

该文提出了一种基于EEMD域统计模型的话音激活检测算法。算法首先利用总体平均经验模态分解(Ensemble Empirical Mode Decomposition,EEMD)对带噪语音进行分解,得到信号的本征模式函数(Intrinsic Mode Function,IMF)分量,选择与原信号的相关性最高的两个分量相加组成主分量;然后对主分量进行频域分解,引入统计模型,求出EEMD域特征参数;最后利用噪声与语音的EEMD域特征参数的不同来进行语音激活检测。实验结果表明,在不同信噪比情况下,本文算法性能优于目前常用的 VAD算法,特别在噪声强度大时体现出明显的优势。     

基于ＥＥＭＤ域统计模型的话音激活检测算法

提出了一种基于EEMD域统计模型的话音激活检测算法。算法首先利用总体平均经验模态分解(Ensemble empirical mode decomposition,EEMD)对带噪语音进行分解,得到信号的本征模式函数(Intrinsicmode function,IMF)分量,选择与原信号的相关性最高的两个分量相加组成主分量;然后对主分量进行频域分解,引入统计模型,求出EEMD域特征参数;最后利用噪声与语音的EEMD域特征参数的不同来进行语音激活检测。实验结果表明,在不同信噪比情况下,本文算法性能优于目前常用的VAD算法,特别在噪声强度大时体现出明显的优势。     

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     

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.     

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。     

Two-microphone separation of speech mixtures.

Separation of speech mixtures, often referred to as the cocktail party problem, has been studied for decades. In many source separation tasks, the separation method is limited by the assumption of at least as many sensors as sources. Further, many methods require that the number of signals within the recorded mixtures be known in advance. In many real-world applications, these limitations are too restrictive. We propose a novel method for underdetermined blind source separation using an instantaneous mixing model which assumes closely spaced microphones. Two source separation techniques have been combined, independent component analysis (ICA) and binary time - frequency (T-F) masking. By estimating binary masks from the outputs of an ICA algorithm, it is possible in an iterative way to extract basis speech signals from a convolutive mixture. The basis signals are afterwards improved by grouping similar signals. Using two microphones, we can separate, in principle, an arbitrary number of mixed speech signals. We show separation results for mixtures with as many as seven speech signals under instantaneous conditions. We also show that the proposed method is applicable to segregate speech signals under reverberant conditions, and we compare our proposed method to another state-of-the-art algorithm. The number of source signals is not assumed to be known in advance and it is possible to maintain the extracted signals as stereo signals.     

基于盲源分离的单通道语音信号增强

在运用基于独立分量分析(ICA)的盲源分离法进行语音增强时,要求观测信号(含噪语音)的个数不少于源信号(纯净语音和噪声)的个数.由于含噪语音通常是单通道的,所以必须合理地生成另一路的虚拟观测信号,以实现纯净语音和噪声的分离是个关键.介绍了一种基于盲源分离和谱减法的单通道语音信号增强的方法.首先运用谱减法对语音进行部分去噪,产生了ICA其中的一路观测信号,并产生了对噪声的估计值.用语音和噪声估计值的帧平均能量构成了加权函数,将噪声的估计值与原始含噪语音进行加权组合,生成另一路的虚拟观测信号.由于虚拟观测信号很好地再现了实际的观测信号,所以运用ICA可以较好地实现了噪声和语音的分离.同时,盲源分离和谱减法相互结合,使语音增强的性能提高.实验证明了算法可以在信噪比很小的情况下实现对噪声的去除,其效果要优于传统的去噪算法.     

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.     

Improved voice activity detection algorithm using wavelet and support vector machine

This paper proposes an improved voice activity detection (VAD) algorithm using wavelet and support vector machine (SVM) for European Telecommunication Standards Institution (ETSI) adaptive multi-rate (AMR) narrow-band (NB) and wide-band (WB) speech codecs. First, based on the wavelet transform, the original IIR filter bank and pitch/tone detector are implemented, respectively, via the wavelet filter bank and the wavelet-based pitch/tone detection algorithm. The wavelet filter bank can divide input speech signal into several frequency bands so that the signal power level at each sub-band can be calculated. In addition, the background noise level can be estimated in each sub-band by using the wavelet de-noising method. The wavelet filter bank is also derived to detect correlated complex signals like music. Then the proposed algorithm can apply SVM to train an optimized non-linear VAD decision rule involving the sub-band power, noise level, pitch period, tone flag, and complex signals warning flag of input speech signals. By the use of the trained SVM, the proposed VAD algorithm can produce more accurate detection results. Various experimental results carried out from the Aurora speech database with different noise conditions show that the proposed algorithm gives considerable VAD performances superior to the AMR-NB VAD Options 1 and 2, and AMR-WB VAD.     

基于curvelet变换和独立分量分析的含噪盲源分离

独立分量分析（ICA）是基于信号高阶统计量的盲源分离方法,在高阶统计量方法中,由于高斯信号的高阶累计量为零,所以系统存在加性高斯噪声时就难以处理。提出了一种基于curvelet阈值去噪和FastICA算法的含噪信号盲分离的方法,并对高斯噪声环境下的混合图像进行了盲分离的仿真。结果表明,该方法能很好地解决由于存在加性高斯噪声而导致经典ICA算法性能发生严重恶化的问题;同时将curvelet变换去噪应用于含噪图像的盲源分离中,可以提高混合图像的信噪比,相对于小波去噪后的ICA算法,其分离性能有很大改善。     

基于Givens变换和二阶振荡W-C-PSO优化的盲源分离算法

针对智能算法在实现盲源分离时容易陷入局部最优且收敛速度缓慢的问题,提出一种基于Givens变换和二阶振荡粒子群优化的盲源分离算法。该算法首先将惯性权重与学习因子两个参数构造函数关系,使之共同调节算法迭代来提高算法的整体性与全局搜索能力;再引入二阶振荡环节增加种群的多样性,这样算法不易陷入局部最优;此外,采用Givens变换将分离矩阵转换成旋转角度表示形式来降低算法的复杂度。仿真表明,该算法能有效实现机械振动信号和语音信号的盲分离,并且相比其他算法具有更快的收敛速度和更好的分离性能。     

Improvement in monaural speech separation using sparse non-negative tucker decomposition

A monaural speech separation/enhancement technique based on non-negative tucker decomposition (NTD) has been introduced in this paper. In the proposed work, the effect of sparsity regularization factor on the separation of mixed signal is included in the generalized cost function of NTD. By using the proposed algorithm, the vector components of both target and mixed signal can be exploited and used for the separation of any monaural mixture. Experiment was done on the monaural data generated by mixing the speech signals from two speakers and, by mixing noise and speech signals using TIMIT and noisex-92 dataset. The separation results are compared with the other existing algorithms in terms of correlation of separated signal with the original signal, signal to distortion ratio, perceptual evaluation of speech quality and short-time objective intelligibility. Further, to get more conclusive information about separation ability, speech recognition using Kaldi toolkit was also performed. The recognition results are compared in terms of word error rate (WER) using the MFCC based features. Results show the average improved WER using proposed algorithm over the nearest performing algorithm is up to 2.7% for mixed speech of two speakers and 1.52% for noisy speech input.     

ICA和线性神经网络在带噪语音分离中的研究

用基于独立分量分析（ICA）的盲源分离方法对强噪声背景下的混合语音信号进行分离时,如果忽略噪声的影响则会产生很差的分离效果。为克服此不足,结合噪声对消和盲源分离,提出了一种在强噪声背景环境下的混合语音分离方法,即先将带噪观测信号通过线性神经网络构成自适应噪声对消器,然后采用ICA进行分离,与增加一路噪声作为源信号的分离方法相比,该方法具有更好的分离效果。     

语音分离与HMM相结合的语音增强方法

针对基于隐马尔科夫（HMM,Hidden Markov Model）的MAP和MMSE两种语音增强算法计算量大且前者不能处理非平稳噪声的问题,借鉴语音分离方法,提出了一种语音分离与HMM相结合的语音增强算法。该算法采用适合处理非平稳噪声的多状态多混合单元HMM,对带噪语音在语音模型和噪声模型下的混合状态进行解码,结合语音分离方法中的最大模型理论进行语音估计,避免了迭代过程和计算量特别大的公式计算,减少了计算复杂度。实验表明,该算法能够有效地去除平稳噪声和非平稳噪声,且感知评价指标PESQ 的得分有明显提高,算法时间也得到有效控制。     

基于Pearson系统模型的盲信号分离研究

独立分量分析是盲源分离的主流技术.自然梯度算法是其中非常重要的算法之一.介绍一种最大似然框架下的Pearson系统模型.该方法的优点是无须知晓信号的概率分布,实验结果表明,该算法能有效地分离随机混合的信号,特别对于非对称源有比同类算法更理想的效果.     

Blind separation of non-stationary sources using continuous density hidden Markov models

Blind source separation (BSS) has attained much attention in signal processing society due to its ‘blind’ property and wide applications. However, there are still some open problems, such as underdetermined BSS, noise BSS. In this paper, we propose a Bayesian approach to improve the separation performance of instantaneous mixtures with non-stationary sources by taking into account the internal organization of the non-stationary sources. Gaussian mixture model (GMM) is used to model the distribution of source signals and the continuous density hidden Markov model (CDHMM) is derived to track the non-stationarity inside the source signals. Source signals can switch between several states such that the separation performance can be significantly improved. An expectation-maximization (EM) algorithm is derived to estimate the mixing coefficients, the CDHMM parameters and the noise covariance. The source signals are recovered via maximum a posteriori (MAP) approach. To ensure the convergence of the proposed algorithm, the proper prior densities, conjugate prior densities, are assigned to estimation coefficients for incorporating the prior information. The initialization scheme for the estimates is also discussed. Systematic simulations are used to illustrate the performance of the proposed algorithm. Simulation results show that the proposed algorithm has more robust separation performance in terms of similarity score in noise environments in comparison with the classical BSS algorithms in determined mixture case. Additionally, since the mixing matrix and the sources are estimated jointly, the proposed EM algorithm also works well in underdetermined case. Furthermore, the proposed algorithm converges quickly with proper initialization.     

两正定矩阵联合对角化盲分离算法

针对具有时间结构的盲分离问题,提出了一种基于两正定矩阵精确联合对角化的盲分离算法。利用多个不同时延统计量构造了两个正定矩阵,以提取出数据的时间结构;再利用所提算法联合对角化构造的两个正定矩阵,得到分离矩阵,进而估计出源信号。所提算法克服了已有算法因采用多个矩阵联合对角化导致的计算量大和采用单个矩阵导致的分离精度低的缺点。计算机仿真结果表明了在有或无噪声情况下,所提算法性能均优于其他对比算法。     

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。     

蚁群聚类的欠定盲源分离方法

利用欠定盲源分离情况下稀疏源信号具有直线聚类的特点,提出了一种估计混叠矩阵的新方法。通过对混叠信号进行标准化处理,使混叠信号形成球形簇,将线性聚类转变成致密聚类;利用蚁群聚类算法对其进行搜索得到聚类中心,从而获得对混叠矩阵的精确估计。该方法能实现源信号数目未知情况下的欠定盲源分离,且能推广到三路或更多路观测信号的情况。对语音信号的仿真结果证明,该方法能精确地分离和恢复原始信号。