Improved variable step-size NLMS adaptive filtering algorithm for acoustic echo cancellation

Acoustic echo canceller (AEC) is used in communication and teleconferencing systems to reduce undesirable echoes resulting from the coupling between the loudspeaker and the microphone. In this paper, we propose an improved variable step-size normalized least mean square (VSS-NLMS) algorithm for acoustic echo cancellation applications based on adaptive filtering. The steady-state error of the NLMS algorithm with a fixed step-size (FSS-NLMS) is very large for a non-stationary input. Variable step-size (VSS) algorithms can be used to decrease this error. The proposed algorithm, named MESVSS-NLMS (mean error sigmoid VSS-NLMS), combines the generalized sigmoid variable step-size NLMS (GSVSS-NLMS) with the ratio of the estimation error to the mean history of the estimation error values. It is shown from single-talk and double-talk scenarios using speech signals from TIMIT database that the proposed algorithm achieves a better performance, more than 3 dB of attenuation in the misalignment evaluation compared to GSVSS-NLMS, non-parametric VSS-NLMS (NPVSS-NLMS) and standard NLMS algorithms for a non-stationary input in noisy environments.     

An efficient stabilized fast Newton adaptive filtering algorithm for stereophonic acoustic echo cancellation SAEC

This paper addresses the field of stereophonic acoustic echo cancellation (SAEC) by adaptive filtering algorithms. Recently, we have proposed a new version of the fast Newton transversal FNTF algorithm for SAEC applications. In this paper, we propose an efficient modification of this algorithm for the same applications. This new algorithm uses a new proposed and simplified numerical stabilization technique and takes into account the cross-correlation between the inputs of the channels. The basic idea is to introduce a small nonlinearity into each channel that has the effect of reducing the inter-channel coherence while not being noticeable for speech due to self masking. The complexity of the proposed algorithm does not alter the complexity of the original version and is kept less than half the complexity of the fastest two-channel FTF filter version. Simulation results and comparisons with the extended two-channel normalized least mean square NLMS and FTF algorithms are presented.     

Performance analysis of under-modelling stereophonic acoustic echo cancellation by adaptive filtering LMS algorithm

This paper addresses the field of stereophonic acoustic echo cancellation (SAEC) with adaptive filtering algorithms. In SAEC applications, using the least mean square (LMS) algorithm, it is usually assumed that the lengths of the adaptive filters are equal to that of the unidentified system responses. Although, in many realistic situations, under-modelled lengths adaptive filters, whose lengths are less than that of the unidentified systems (under-modelled systems), are employed, and analysis results for the exact modelled stereophonic LMS algorithm are not automatically appropriate to the under-modeled lengths. In this paper, we present a statistical analysis of the under-modeled stereophonic LMS algorithm. Exact expressions and deterministic recursive equations to the mean coefficients behavior of the adaptive LMS filters are derived to completely characterize and assess the performances (transient and steady-state) of the under-modeling stereophonic LMS algorithm. The expected theoretical behaviour is compared with Monte Carlo simulations and practical experimental results, showing a very good agreement.     

Stereophonic acoustic echo cancellation employing selective-tap adaptive algorithms

Stereophonic acoustic echo cancellation has generated much interest in recent years due to the nonuniqueness and misalignment problems that are caused by the strong interchannel signal coherence. In this paper, we introduce a novel adaptive filtering approach to reduce interchannel coherence which is based on a selective-tap updating procedure. This tap-selection technique is then applied to the normalized least-mean-square, affine projection and recursive least squares algorithms for stereophonic acoustic echo cancellation. Simulation results for the proposed algorithms have shown a significant improvement in convergence rate compared with existing techniques.     

Low complexity LMS-type adaptive algorithm with selective coefficient update for stereophonic acoustic echo cancellation

Stereophonic acoustic echo cancellation (SAEC) has brought up recently much attention and found a viable place in a number of hands-free applications. In this paper, we propose an LMS-type algorithm for SAEC based on decomposing the long adaptive filter of each channel of the SAEC system into smaller subfilters. We further reduce the complexity of the algorithm by employing the selective coefficient update (SCU) method in each subfilter. This leads to a significant improvement in the convergence rate of the algorithm with low computational overhead. However, the algorithm has a high final mean-square error (MSE) at steady-state that increases as number of subfilters increases. A combined-error algorithm is presented that achieves fast convergence without compromising the steady state error level. Simulations demonstrate the convergence speed advantages of the combined-error algorithm.     

一种新的基于参数估计的自适应双边滤波算法*

传统双边滤波算法需要根据经验预先设置空间标准差和灰度标准差,参数固定且不具有通用性。针对此问题,提出了一种新的基于参数估计的自适应双边滤波算法。通过图像灰度共生矩阵实现空间标准差的自适应,利用统计方法估计光滑区域噪声标准差,根据噪声标准差设置灰度标准差,从而实现自适应双边滤波。仿真结果表明,所提出的算法无论在主观还是客观评价上都取得了较好的效果。     

基于声回波抵消两路算法的改进更新逻辑

声回波抵消两路算法被广泛用来检测系统双向通话;基于声回波抵消两路算法,提出了一种改进的控制更新逻辑。此更新逻辑通过比较滤波器的回波返回损失(ERLE),判断是否对滤波器进行更新。此改进更新逻辑能正确检测系统双向通话,避免滤波器的错误更新,并提高两路算法的收敛速度,减小存储器资源和计算量。仿真结果证实了此更新逻辑的有效性。     

A new class of adaptive algorithm based on correlation functions for double-talk acoustic echo cancelling

Echo path estimation in echo cancelling for teleconference system is a problem in double talk condition. The correlation processing algorithm was defined by the authors to solve this problem. In this paper, a new fast implementation in the frequency domain with correlation initialization and step size normalization is introduced. The new frequency-domain ECLMS (FECLMS) algorithm requires only 2.7% of the ECLMS computational loads. The new algorithm is superior as compared with conventional one in double-talk with speech signal.     

变步长比例归一化子带自适应滤波算法研究

针对欠定模型条件下定步长比例归一化子带自适应滤波(PNSAF)算法收敛速度和稳态误差之间的矛盾,提出了一种变步长VSS-PNSAF算法。该算法将系统干扰噪声和欠定模型噪声对系统性能的影响考虑进滤波器系数更新过程中,利用后验误差对其进行补偿,根据先验误差与后验误差之间的联系,导出了一种适用于比例归一化子带自适应滤波算法的步长调节方法。该算法综合了子带自适应滤波、比例自适应算法及变步长方法的优点。仿真结果表明:与定步长比例归一化子带自适应滤波算法相比,所提算法具有更低的稳态误差和更快的收敛速度。     

基于自适应特性二维经验模式分解的Retinex彩色图像增强

提出一种彩色图像自适应增强方法：将图像从RGB色彩空间转化到HSV色彩空间并保持H分量不变,对亮度分量V通过自适应特性二维经验模式分解（ABEMD）估算其照度分量,再根据中心/环绕Retinex算法计算出反射分量,对照度和反射分量分别应用Gamma校正和Weber定律,并进行加权运算,基于全局特性自适应地调整S分量,并将图像从HSV色彩空间转化回RGB色彩空间。最后利用主观和客观的方法对实验结果进行了评价,实验表明了该算法在均值、方差、信息熵和清晰度方面均优于MSR算法和Meylan的算法。     

基于二维经验模态和均值滤波的图像去噪方法

提出了一种新的图像去噪方法。此方法通过二维经验模态（BEMD）将噪声图像分解为一系列不同频带的子图像。对低频近似图像保持不变,对高频细节图像采用不同的模板进行均值滤波,最后将低频近似图像和均值滤波后的图像合成为去噪后的图像。实验结果表明该方法在滤除图像噪声的同时,又能较好地保留图像的边缘细节,其滤波效果优于单一的BEMD图像去噪和均值滤波图像去噪以及小波变换和均值滤波图像去噪方法。     

具有双方对讲保护的自适应回波消除新算法

基于电话会议的声学回波中的双方对讲情况,本文提出了一个无需设置双方对讲检测器,但仍能在双讲过程中保护自适应滤波器消除性能的NLMS类算法.由于可以由远端信号和近端混合接收信号之间的相关性系数的变化来判断双讲发生或回波路径改变,所以改进的算法中直接将其代入滤波器权系数的迭代公式中,从而控制滤波器系数更新的快慢.仿真结果表明与同类算法相比,采用更小的计算量,该算法在双方对讲时能较好地起到保护作用,而在回波路径改变时也具有快速的跟踪性能.     

A new robust double-talk detector based on the Stockwell transform for acoustic echo cancellation

Despite great developments in the field of acoustic echo cancellation (AEC), the presence of double-talk remains difficult problem. The main role of double-talk detection (DTD) is to control adaptation of the filter coefficients by halting their update in double-talk situations. In this paper, we propose a new method of DTD based on a time–frequency analysis that uses the Stockwell transform (ST).The ST is a time–frequency spectral localization method that combines the characteristics of the short-time Fourier transform and the wavelet transform. This method provides better time–frequency resolution, especially for non-stationary signals. In the experimental tests, the normalized least mean squares (NLMS) algorithm is used to update the filter coefficients along with speech signals taken from the TIMIT database. The obtained results show better performance compared to existing methods in terms of misalignment convergence and speech intelligibility enhancement.     

A novel adaptive lattice filtering algorithm for alpha-stable processes

This paper generalizes Burg's formula to be suitable for alpha-stable distribution and proposes a novel adaptive lattice algorithm for adaptive parameter estimation of alpha-stable AR processes. The performance is compared to that of other algorithms with lattice structure for alpha-stable AR processes for different α parameter. Simulations studies indicate that the proposed algorithm shows superior convergence speed over existing well-known lattice algorithms in parameter estimation of alpha-stable processes.     

基于经验模态分解的算法改进

经验模态分解（EMD）算法是Hilbert—Huang变换（HHT）的核心算法,它的分解效果依赖于采样频率的选择,介绍一种新的EMD的采样频率选取方法,并通过仿真信号实验表明该方法分解信号更完全,对电力系统谐波检测分析有一定的实际应用价值。     

基于经验模态分解差分谱的故障选线新方法

提出了一种基于经验模态分解差分谱的小电流接地系统单相接地故障选线新方法。该方法采用经验模态分解方法分解故障发生后各条线路的暂态零序电流,提取出各条线路的高频IMF1分量,并根据差分谱和波动值的定义,求出各条线路的IMF1分量差分谱及IMF1分量波动值;通过比较各条线路的IMF1分量波动值,得出IMF1分量波动值最大的线路,则该线路即为故障线路。仿真结果表明,该方法具有较高的选线准确度,适用于不同接地电阻与不同故障初相角情况下的小电流接地系统故障选线。     

基于二次二维经验模态分解去噪的湍流退化图像复原算法

为实现大气湍流环境下的高质量成像,将自适应光学波前探测技术与数字图像处理技术相结合,并提出了一种基于二次二维经验模态分解去噪的湍流退化图像复原算法。通过在光学系统中使用哈特曼—夏克波前传感器探测波前信息,进而计算光学系统点扩散函数;然后使用改进的二次二维经验模态分解算法进行图像去噪,最后利用R-L算法实现对湍流退化图像的复原。通过搭建光学实验系统,对实际拍摄的湍流退化图像进行了复原实验。结果表明,该算法能够有效减弱噪声放大现象,得到更加稳定的高质量大气湍流退化图像复原结果。     

基于矩阵分解的协同过滤算法

协同过滤推荐算法是电子商务推荐系统中运用最成功的一种推荐技术。针对目前大多数协同过滤算法普遍存在的可扩展性和抗稀疏性问题,在传统的矩阵分解模型（SVD）的基础上提出了一种带正则化的基于迭代最小二乘法的协同过滤算法。通过对传统的矩阵分解模型进行正则化约束来防止模型过度拟合训练数据,并通过迭代最小二乘法来训练分解模型。在真实的实验数据集上实验验证,该算法无论是在可扩展性,还是在抗稀疏性方面均优于几个经典的协同过滤推荐算法。     

An improved new error estimation algorithm for optimal filter lengths for stereophonic acoustic echo cancellation

In this paper, we propose an new error estimate algorithm (NEEA) for stereophonic acoustic echo cancellation (SAEC) that is based on the error estimation algorithm (EEA) in [Nguyen-Ky T, Leis J, Xiang W. An improved error estimate algorithm for stereophonic acoustic echo cancellation system. In: International conference on signal processing and communication systems, ICSPCS’2007, Australia; December 2007]. In the EEA and NEEA, with the minimum error signal fixed, we compute the filter lengths so that the error signal may approximate the minimum error signal. When the echo paths change, the adaptive filter automatically adjusts the filter lengths to the optimum values. We also investigate the difference between the adaptive filter lengths. In contrast with the conclusions in [Khong AWH, Naylor PA. Stereophonic acoustic echo cancellation employing selective-tap adaptive algorithms. IEEE Trans Audio, Speech, Lang Process 2006;14(3):785-96, Gansler T, Benesty J. Stereophonic acoustic echo cancellation and two channel adaptive filtering: an overview. Int J Adapt Control Signal Process 2000;4:565-86, Benesty J, Gansler T. A multichannel acoustic echo canceler double-talk detector based on a normalized cross-correlation matrix. Acoust Echo Noise Control 2002;13(2):95-101, Gansler T, Benesty J. A frequency-domain double-talk detector based on a normalized cross-correlation vector. Signal Process 2001;81:1783-7, Eneroth P, Gay SL, Gansler T, Benesty J. A real-time implementation of a stereophonic acoustic echo canceler. IEEE Trans. Speech Audio Process 2001;9(5):513-23, Gansler T, Benesty J. New insights into the stereophonic acoustic echo cancellation problem and an adaptive nonlinearity solution. IEEE Trans. Speech Audio Process 2002; 10(5):257-67, Benesty J, Gansler T, Morgan DR, Sondhi MM, Gay SL. Advances in network and acoustic echo cancellation. Berlin: Springer-Verlag; 2001], our simulation results have shown that the filter lengths can be different. Our simulation results also confirm that the NEEA is better than EEA and SM-NLMS algorithm in terms of echo return loss enhancement.     

MC-CDMA系统中基于QR分解的多址干扰消除算法

多址干扰是导致MC-CDMA系统误码性能下降的重要因素。为了消除多址干扰,提出基于接收信号功率排序和QR分解的多址干扰消除算法（Power-based MMSE Sorted QR Decomposition,P-based MMSE-SQRD）对MC-CDMA系统的上行链路进行检测。仿真结果显示,与传统的串行干扰消除算法相比,P-based MMSE-SQRD的误码性能有明显的提高,而且其复杂度也相对较低。