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.     

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。     

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。     

混合式自适应Kalman滤波算法

采用虚拟噪声补偿模型误差和有偏的噪声方差估值器、滤波器收敛性判据相结合的方法来解决自适应Kalman滤波发散的问题。首先若模型不准确,则引入虚拟噪声对模型误差进行虚拟补偿,然后采用有偏的噪声方差估值器、滤波器收敛性判据对噪声方差估计值进行监控,阻止滤波器发散。采用混合式自适应Kalman滤波算法对Gill公司的风向风速仪实时采集的数据进行处理,实验结果表明,该方法能有效的提高性能、抑制滤波发散,具有较强的实用性、自适应能力。     

基于改进NLMS算法的自适应滤波算法

最小均方算法是应用最广泛的自适应算法之一,但其收敛速度欠佳。在传统NLMS算法的基础上,提出了重复调整归一化最小均方算法(DRNLMS)即在相邻两输入信号样本的间隔时间进行额外调整运算,以提高算法的收敛性,并通过计算机仿真实现该算法。     

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.     

A sequential learning algorithm based on adaptive particle filtering for RBF networks

To address the problem of low filtering accuracy and divergence caused by unknown process noise statistics and local linearization in neural network state-space model, this paper proposes an adaptive process noise covariance particle filter algorithm for the radial basis function (RBF) networks. Using the algorithm, the evolution of the weights and centers of RBF networks is achieved sequentially in time by use of the extended Kalman particle filter algorithm, and the process noise covariance matrices are also obtained simultaneously by maximizing the evidence density function with respect to the process noise covariance matrices. Performance of the presented approach is evaluated by two function approximation problems. Experimental results show that the proposed approach obtains better prediction accuracy than other well-known training algorithms.     

Hybrid adaptive evolutionary algorithm based on decomposition

The performance of search operators varies across the different stages of the search/optimization process of evolutionary algorithms (EAs). In general, a single search operator may not do well in all these stages when dealing with different optimization and search problems. To mitigate this, adaptive search operator schemes have been introduced. The idea is that when a search operator hits a difficult patch (under-performs) in the search space, the EA scheme “reacts” to that by potentially calling upon a different search operator. Hence, several multiple-search operator schemes have been proposed and employed within EA. In this paper, a hybrid adaptive evolutionary algorithm based on decomposition (HAEA/D) that employs four different crossover operators is suggested. Its performance has been evaluated on the well-known IEEE CEC’09 test instances. HAEA/D has generated promising results which compare well against several well-known algorithms including MOEA/D, on a number of metrics such as the inverted generational distance (IGD), the hyper-volume, the Gamma and Delta functions. These results are included and discussed in this paper.     

基于二维EMD的纹理分类方法

在纹理图像的二维经验模式分解的筛分过程中,使用Delaunay三角形网格和三次多项式插值来构造筛分过程的包络,然后提取样本的每个内蕴模函数中过零点数目以及极值处振幅的均值作为特征向量来进行训练,根据训练得到的样本特征数据库对纹理图像进行分类.实验证明,使用所选取的特征对纹理图像进行分类的方法是可行的.     

自适应双边滤波的Retinex图像增强算法

针对现有的Retinex算法不能自动调节参数,提出一种基于参数估计的双边滤波Retinex算法。该算法首先利用主成份分析和Canny边缘检测算法分别进行噪声估计和边缘强度估计;然后通过线性相关运算计算双边滤波的空间几何标准差参数和亮度标准差参数;再利用参数估计的双边滤波把图像分解出照度图像和反射图像;最后将照度图像和反射图像通过不同方法的压缩和增强并合成一幅新的图像。通过实验表明,它不仅能够自动设置参数,还能有效抑制光晕现象。     

Efficient blind dereverberation and echo cancellation based on independent component analysis for actual acoustic signals

This letter presents a new algorithm for blind dereverberation and echo cancellation based on independent component analysis (ICA) for actual acoustic signals. We focus on frequency domain ICA (FD-ICA) because its computational cost and speed of learning convergence are sufficiently reasonable for practical applications such as hands-free speech recognition. In applying conventional FD-ICA as a preprocessing of automatic speech recognition in noisy environments, one of the most critical problems is how to cope with reverberations. To extract a clean signal from the reverberant observation, we model the separation process in the short-time Fourier transform domain and apply the multiple input/output inverse-filtering theorem (MINT) to the FD-ICA separation model. A naive implementation of this method is computationally expensive, because its time complexity is the second order of reverberation time. Therefore, the main issue in dereverberation is to reduce the high computational cost of ICA. In this letter, we reduce the computational complexity to the linear order of the reverberation time by using two techniques: (1) a separation model based on the independence of delayed observed signals with MINT and (2) spatial sphering for preprocessing. Experiments show that the computational cost grows in proportion to the linear order of the reverberation time and that our method improves the word correctness of automatic speech recognition by 10 to 20 points in a RT??= 670 ms reverberant environment.     

A new adaptive filtering subband algorithm for two-channel acoustic noise reduction and speech enhancement

This paper addresses the problem of acoustic noise reduction and speech enhancement by adaptive filtering algorithms. Most speech enhancement methods and algorithms which use adaptive filtering structure are generally expressed in fullband form. One of these widespread structures is the Forward Blind Source Separation Structure (FBSS). This FBSS structure is often used to separate speech form noise and therefore enhance the speech signal at the processing output. In this paper, we propose a new subband implementation of this FBSS structure. In order to give more robustness to the proposed structure, we adapt then we apply to this subband structure a new combination of criteria based on the system mismatch and the smoothing filtering errors minimizations. The combination between this proposed subband structure with this optimal criteria allows to obtain a new two-channel subband forward (2CSF) algorithm that improves the convergence speed of the cross adaptive filters which are used to separate speech from noise. Objective tests under various environments are presented showing the good behavior of the proposed 2CSF algorithm.     

Packet based echo cancellation for VoIP networks

In today’s modern telephony network, VoIP is fast emerging as one of the main communication techniques. However, the performance and the quality of VoIP are affected by echo. Packet Based Echo Canceller (PBEC) is introduced, as a solution to cancel echo in the VoIP network. PBEC can replace the current echo cancellers, which are located in the Public Switched Telephony Network (PSTN) central switches. The operating principle of the PBEC is explained and its advantages are highlighted. The performance of the PBEC using different speech codecs is also studied. Using the PBEC, a maximum Echo Return Loss Enhancement (ERLE) of 37.39 dB has been achieved when used with the Pulse Code Modulation (PCM) based speech codec. From the simulation results, it can be seen that the performance of the Adaptive Differential Pulse Code Modulation (ADPCM) clearly matches the performance of the PCM based speech codec. The other major problem affecting the VoIP network is the issue of packet loss. This issue of packet loss has been successfully addressed in this paper by the insertion of random values. With the insertion of random values, the ERLE increases by 4.81 dB compared to when there is no insertion of random value. The PBEC with the utilization of random values would make the VoIP a better communication tool.     

基于快速二维经验模态分解的纹理分割

经验模态分解是一种数据驱动的信号分解方法,具有局部性和瞬时性等特性,非常适合非稳态非线性信号分析.提出了一种新的快速二维经验模态分解方法,在新方法中,采用了新的边界抑制算法,改进了经验模态分解算法的筛选条件.将该方法应用于纹理分割,取得了满意的实验效果.     

一种基于卡尔曼滤波的队列长度自适应算法

传统主动队列管理(AQM)算法在处理传感器网络突发流时具有响应速度慢、抗网络突变性能弱的缺点.针对此问题,提出了一种新的AQM算法,算法首先将队列长度作为早期拥塞检测参量,运用卡尔曼滤波理论预测队列长度;其次根据队列长度在缓冲区的占用比来划分网络状态;最后根据不同占用比采取相应的丢包策略,自适应地调整丢包率,当出现网络突变时,加大调整幅度,使队列长度保持在理想区间.仿真实验表明:新算法能够较好地适应网络波动,提高网络服务质量(QoS),算法综合性能优于主流AQM算法.     

基于极大似然估计的新息自适应滤波算法

针对噪声统计信息未知或时变情况下常规卡尔曼滤波估计精度下降甚至发散的问题,提出了一种基于极大似然估计的新息自适应滤波算法.算法对基于极大似然估计的常规新息协方差估值器进行限定记忆指数衰减加权修正,增加滑动窗口内新近新息协方差序列的利用权重;根据新息自适应原理,利用新息协方差估计值直接计算滤波增益矩阵,加快滤波器收敛速度的同时提高了滤波算法的估计精度.算法应用于捷联惯性导航系统/全球定位系统(SINS/GPS)组合导航系统,仿真实验表明:在噪声统计信息未知或时变情况下,算法具有更强的鲁棒性以及更高的滤波精度.