基于SOR-PCG的低复杂度信号检测算法研究

针对最小均方误差信号检测算法复杂度随着天线数量增加呈指数增长的问题,提出低复杂度的预处理共轭梯度信号检测算法.该算法通过灵活调整松弛因子,运用预处理技术降低矩阵条件数,从而加快共轭梯度信号检测算法的收敛速度.仿真结果显示,该算法在小数量的迭代中能够达到和最小均方误差检测算法相似的误码率,算法复杂度下降了一个数量级.通过选择适当的松弛因子,相比直接用共轭梯度法,能够更快收敛到最佳值.     

On the Performance Analysis of a Class of Transform-domain NLMS Algorithms with Gaussian Inputs and Mixture Gaussian Additive Noise Environment

This paper studies the convergence performance of the transform domain normalized least mean square (TDNLMS) algorithm with general nonlinearity and the transform domain normalized least mean M-estimate (TDNLMM) algorithm in Gaussian inputs and additive Gaussian and impulsive noise environment. The TDNLMM algorithm, which is derived from robust M-estimation, has the advantage of improved performance over the conventional TDNLMS algorithm in combating impulsive noises. Using Price’s theorem and its extension, the above algorithms can be treated in a single framework respectively for Gaussian and impulsive noise environments. Further, by introducing new special integral functions, related expectations can be evaluated so as to obtain decoupled difference equations which describe the mean and mean square behaviors of the TDNLMS and TDNLMM algorithms. These analytical results reveal the advantages of the TDNLMM algorithm in impulsive noise environment, and are in good agreement with computer simulation results.     

基于正交小波变换的前馈神经网络盲均衡算法

针对前馈神经网络（FNN）盲均衡算法收敛速度慢、均方误差大的缺点,在分析FNN盲均衡算法和正交小波变换（OWT）理论的基础上,提出了基于正交小波变换的FNN盲均衡算法。该算法利用正交小波变换良好的去相关性,对FNN均衡器输入信号进行预处理后,降低了输入信号的自相关性,从而加快了收敛速度和减小了均方误差。水声信道盲均衡的仿真结果表明,该算法在收敛速度与均方误差方面的性能比FNN盲均衡算法优越。     

基于基扩展模型的改进正则化正交匹配追踪V2X 快时变SC-FDMA信道估计

为了进一步提升车联万物(V2X)的通信性能,首先根据信道冲激响应的稀疏性建立了适用于高速移动场景的基扩展模型(BEM);其次,证明了BEM系数具有稀疏性,将信道估计问题转化为稀疏信号重构问题,进而提出基于BEM的改进正则化正交匹配追踪(iROMP)迭代稀疏信道估计算法(简称为BEM-iROMP算法)。所提算法通过iROMP获取BEM系数,利用反馈结果不断迭代以达到最优信道估计。仿真结果表明,与最小二乘法、线性最小均方误差和BEM-LS信道估计算法相比,所提算法能够有效提高V2X快时变信道下单载波频分多址系统的归一化均方误差和误码率性能。     

Split Wiener filtering with application in adaptive systems

This paper proposes a new structure for split transversal filtering and introduces the optimum split Wiener filter. The approach consists of combining the idea of split filtering with a linearly constrained optimization scheme. Furthermore, a continued split procedure, which leads to a multisplit filter structure, is considered. It is shown that the multisplit transform is not an input whitening transformation. Instead, it increases the diagonalization factor of the input signal correlation matrix without affecting its eigenvalue spread. A power normalized, time-varying step-size least mean square (LMS) algorithm, which exploits the nature of the transformed input correlation matrix, is proposed for updating the adaptive filter coefficients. The multisplit approach is extended to linear-phase adaptive filtering and linear prediction. The optimum symmetric and antisymmetric linear-phase Wiener filters are presented. Simulation results enable us to evaluate the performance of the multisplit LMS algorithm.     

鲁棒总体均方最小自适应滤波:算法与分析

本文研究了在输入输出观测数据均含有噪声的情况下如何有效地进行鲁棒自适应滤波的问题.以总体均方误差(TMSE)最小为准则,基于最速下降原理,通过对总体均方误差梯度进行修正,提出了一种鲁棒的总体均方最小自适应滤波算法.通过与已有算法的对比分析表明,该算法能够有效地降低权向量的每步调整量对噪声的敏感程度.仿真实验的结果进一步表明,该算法的鲁棒抗噪性能和稳态收敛精度明显地高于其它同类方法,而且可以使用较大的学习因子,在高噪声环境下仍然保持良好的收敛性.     

一种约束稳定性最小均方噪声对消算法

研究了最小均方误差(LMS)算法、归一化的最小均方(NLMS)算法及变步长NLMS算法在自适应噪声干扰抵消器中的应用,针对目前这些算法在噪声对消器应用中的缺点,将约束稳定性最小均方(CS-LMS)算法应用到噪声处理中,并进一步结合变步长的思想提出来一种新的变步长CS-LMS算法。通过MATLAB进行仿真分析,结果证实提出的算法与其他算法相比,能很好地滤除掉噪声从而得到期望信号,明显的降低了稳态误差,并拥有好的收敛速度。     

On the convergence behavior of the LMS and the normalized LMSalgorithms

It is shown that the normalized least mean square (NLMS) algorithm is a potentially faster converging algorithm compared to the LMS algorithm where the design of the adaptive filter is based on the usually quite limited knowledge of its input signal statistics. A very simple model for the input signal vectors that greatly simplifies analysis of the convergence behavior of the LMS and NLMS algorithms is proposed. Using this model, answers can be obtained to questions for which no answers are currently available using other (perhaps more realistic) models. Examples are given to illustrate that even quantitatively, the answers obtained can be good approximations. It is emphasized that the convergence of the NLMS algorithm can be speeded up significantly by employing a time-varying step size. The optimal step-size sequence can be specified a priori for the case of a white input signal with arbitrary distribution     

基于零阶统计量的韧性归一化自适应时间延迟估计

依据零阶统计量理论,给出对数矩过程、对数宽平稳及对数各态遍历的定义,提出一种韧性的归一化自适应时间延迟估计方法(简称NZOSTDE).该算法用FIR滤波器对两个含有脉冲噪声的观测信号建模,利用不存在有限方差的脉冲信号经过对数变换后其各阶矩的存在性和几何功率的概念,在对数域基于最小均方误差(LMS)准则归一化自适应得到FIR滤波器的系数,该系数最大值对应的序号就是时间延迟的估计值.本文提出的新算法克服了基于分数低阶统计量(FLOS)算法的局限性.计算机仿真实验表明,NZOSTDE算法在强脉冲噪声环境下比归一化最小平均P范数时间延迟估计方法(简称NLMPTDE)算法更具有韧性.     

有记忆效应的功放实数延时模糊神经网络模型

描述了一种基于实数延时模糊神经网络的有记忆效应的功率放大器模型.该模糊神经系统即自适应模糊神经推理系统,采用模糊c类均值聚类方法来减少模型的规则数目和简化模型结构.在训练过程中,采用最小二乘和反向传播相结合的高效算法提取模型参数.在测试平台上用三载波WCDMA宽带信号对射频功率放大器进行测试,并借助矢量信号分析仪采样功率放大器输入和输出数据,成功地对模型进行了训练和验证.通过和实数延时神经网络模型(RVTDNN)比较,该模型的收敛速度远快于这些前馈结构的神经网络模型.比较和分析时域和频域结果表明模型有很好的性能,其归一化均方误差达-38dB.     

Adaptive Beamformers for High-Speed Mobile Communication

In this work, we propose a fast conjugate gradient method (CGM) for beamforming, after thoroughly analyzing the performances of the least mean square (LMS), the recursive least square (RLS), and the sample matrix inversion (SMI) adaptive beamforming algorithms. Various experiments are carried out to analyze the performances of each beamformer in detail. The proposed conjugate gradient method does not use the Eigen spread of the signal correlation matrix as in the case of the LMS and the RLS methods. It computes antenna array weights orthogonally for each iteration. Hence the convergence rate and the null depths of the proposed method are much better than the LMS, the SMI the RLS and the classical CGM. Also, the simulation results confirm that this method has a speed improvement of about 60% over the classical conjugate gradient method. This aspect significantly reduces the processor burden and saves a lot of power during the beamforming process. Hence the proposed method is superior compared to the LMS, the RLS, the SMI, and classical CGM and most suitable for high-speed mobile communication.

     

Sparse adaptive constant blind equalization algorithm for sparse multipath channel

In order to improve the convergence rate of the blind equalizer for sparse multipath channel,a novel blind equalization approach called l₀-norm constraint proportionate normalized least mean square constant algorithm was proposed for M-order phase-shift keying (MPSK) signal.Based on the constant modulus characteristics of MPSK signal and the sparse property of equalizer,a new blind equalization cost function with the l₀-norm penalty on the equalizer tap coefficients was firstly constructed.Then the update formula of the tap coefficients was derived according to the gradient descent algorithm.Moreover,the iteration step was updated by drawing upon the normalized proportionate factor.The algorithm not only assigned step sizes proportionate to the magnitude of the current individual tap weights,but also attracted the inactive taps to zero adaptively.Theoretical analysis and simulation results show that the proposed algorithm outperforms the existing blind equalization algorithms for sparse channel in reducing ISI and improving convergence rate.     

Improved approximate QR-LS algorithms for adaptive filtering

This paper studies a class of O(N) approximate QR-based least squares (A-QR-LS) algorithm recently proposed by Liu in 1995. It is shown that the A-QR-LS algorithm is equivalent to a normalized LMS algorithm with time-varying stepsizes and element-wise normalization of the input signal vector. It reduces to the QR-LMS algorithm proposed by Liu et al. in 1998, when all the normalization constants are chosen as the Euclidean norm of the input signal vector. An improved transform-domain approximate QR-LS (TA-QR-LS) algorithm, where the input signal vector is first approximately decorrelated by some unitary transformations before the normalization, is proposed to improve its convergence for highly correlated signals. The mean weight vectors of the algorithms are shown to converge to the optimal Wiener solution if the weighting factor w of the algorithm is chosen between 0 and 1. New Givens rotations-based algorithms for the A-QR-LS, TA-QR-LS, and the QR-LMS algorithms are proposed to reduce their arithmetic complexities. This reduces the arithmetic complexity by a factor of 2, and allows square root-free versions of the algorithms be developed. The performances of the various algorithms are evaluated through computer simulation of a system identification problem and an acoustic echo canceller.     

Parametric modeling of somatosensory evoked potentials usingdiscrete cosine transform

This paper introduces a parametric method for identifying the somatosensory evoked potentials (SEPs). The identification was carried out by using pole-zero modeling of the SEPs in the discrete cosine transform (DCT) domain. It was found that the DCT coefficients of a monophasic signal can be sufficiently approximated by a second-order transfer function with a conjugate pole pair. The averaged SEP signal was modeled by the sum of several second-order transfer functions with appropriate zeros and poles estimated using the least square method in the DCT domain. Results of the estimation demonstrated that the model output was in an excellent agreement with the raw SEPs both qualitatively and quantitatively. Comparing with the common autoregressive model with exogenous input modeling in the time domain, the DCT domain modeling achieves a high goodness of fitting with a very low model order. Applications of the proposed method are possible in clinical practice for feature extraction, noise cancellation and individual component decomposition of the SEPs as well as other evoked potentials.     

基于小波变换变步长LMS自适应信道均衡的新方法

结合变换域最小均方(LMS)和变步长LMS算法的优势,提出了一种基于小波变换的变步长LMS自适应均衡方法。该方法中步长调整函数采用了改进的Sigmoid函数,该函数具有简单且误差信号接近零时变化缓慢的特点。并且,在训练模式、判决引导模式以及混合模式下,将提出方法和传统均衡方法进行了仿真比较。结果表明,所提出的方法比传统的线性LMS算法、变步长LMS以及小波变换LMS收敛更快、性能更优。     

基于离散Chirp—Fourier变换的线性调频信号的Wiener滤波

针对白噪声中线性调频信号的滤波问题 ,提出了一种基于Chirp Fourier变换的Wiener滤波算法 ,根据线性变换等效Wiener滤波的原理 ,利用最小二乘法导出了Chirp Fourier域上的Wiener滤波算子。理论分析及仿真结果表明 ,该算法不仅能够给出信号波形的最小均方误差估计 ,还可利用FFT实现 ,且实现较为简便。     

Robust Regularization for Normalized LMS Algorithms

We present a novel normalized least mean square (NLMS) algorithm with robust regularization. The proposed algorithm dynamically updates the regularization parameter that is fixed in the conventional$epsilon $-NLMS algorithms. By exploiting the gradient descent direction we derive a computationally efficient and robust update scheme for the regularization parameter. Through experiments we demonstrate that the proposed algorithm outperforms conventional NLMS algorithms in terms of the convergence rate and the misadjustment error.     

A new optimized least-square sparse channel estimation scheme for underwater acoustic communication

In underwater acoustic (UWA) communication, orthogonal frequency division multiplexing (OFDM) is a promising technology that is highly essential to get channel state information meant for channel estimation (CE). Nevertheless, higher complexity, slower convergence, and poor performance, which degrade the performance estimation, are the limitations of the traditional CE methodologies. Thus, by amalgamating the least square (LS)-CE algorithm along with polynomial interpolated black widow optimization (PI-BWO) model, an optimized least square sparse (OLSS) CE algorithm has been proposed to intend for a UWA-OFDM communication system. Formerly, by utilizing the 2's complement shift left turbo encoding (2CSL-TE) methodology, the input signal is encoded. After that, the modulated encoded signal is provided for inverse fast Fourier transform (IFFT) operations; subsequently, they are transferred over the UWA channel toward the receiver OFDM. By employing the OLSS methodology, the received OFDM signal's interference-free region is utilized for sparse CE at the receiver. Regarding symbol error rate (SER), bit error rate (BER), mean square error (MSE), and peak signal-to-noise ratio (PSNR), the proposed model's experiential outcome is evaluated and analogized with the other prevailing methodologies. When analogized with the conventional models, the proposed estimation methodologies achieved better performance.     

Analysis of LMS and NLMS algorithms with delayed coefficient updateunder the presence of spherically invariant processes

Certain conditions require a delay in the coefficient update of the least mean square (LMS) and normalized least mean square (NLMS) algorithms. This paper presents an in-depth analysis of these modificated versions for the important case of spherically invariant random processes (SIRPs), which are known as an excellent model for speech signals. Some derived bounds and the predicted dynamic behavior of the algorithms are found to correspond very well to simulation results and a real time implementation on a fixed-point signal processor. A modification of the algorithm is proposed to assure the well known properties of the LMS and NLMS algorithms     

Least square identification of alias components of linear periodically time-varying systems and optimal training signal design

A least-squares (LS) method for identifying alias components of discrete linear periodically time-varying (LPTV) systems is proposed.The authors apply a periodic input signal to a finite impulse response (FIR)--LPTV system and measure the noise-contaminated output.The output of this LPTV system has the same period as the input when the period of the input signal is amultiple of the period of the LPTV system.The authors show that the input and the output can be related by using the discrete Fourier transform. In the frequency domain, an LS method can be used to identify the alias components. A lower bound on the mean square error (MSE) of the estimated alias components is given for FIR--LPTV systems.The optimal training signal achieving this lower MSE bound is designed subsequently. The algorithm is extended to the identification of infinite impulse response (IIR)--LPTV systems as well. Simulation results show the accuracy of the estimation and the efficiency of the optimal training signal design.