Modified leaky LMS algorithm for channel estimation in DS-CDMAsystems

A simple adaptive least mean square (LMS) type algorithm for channel estimation is developed based on certain modifications to finite-impulse response (FIR) Wiener filtering. The proposed algorithm is nearly blind since it does not require any training sequence or channel statistics, and it can be implemented using only noise variance knowledge. A condition guaranteeing the convergence of the algorithm and theoretical mean square error (MSE) values are also derived. Computer simulation results demonstrate that the proposed algorithm can yield a smaller MSE than existing techniques, and that its performance is close to that of optimal Wiener filtering     

Two new algorithms for tracking arterial parameters innonstationary noise conditions

Two new algorithms with reduced sensitivity to the changing environment are applied to tracking arterial circulation parameters. They are variants of the Least-Squares (LS) algorithm with Variable Forgetting factor (LSVF), and of the Constant Forgetting factor-Covariance Modification (CFCM) LS algorithm, devised to overcome their main practical deficiencies related to noise level sensitivity and the high number of design variables, respectively. To this end, adaptive mechanisms are incorporated to estimate observation noise variance in LSVF and the rate of change for the different parameters in CFCM. Specific computer simulation experiments are presented to compare their effectiveness with the original counterparts and to provide guidelines for their optimal tuning at different noise levels. Moreover, algorithm performance degradation, consequent on changes in the noise level compared to that assumed during the tuning phase, is analyzed. In particular, it is shown that, when the noise level changes with respect to the tuning value, the new LSVF algorithm is much more robust than the original one, whose performance degrades rapidly. The new CFCM algorithm is characterized by a reduced number of design variables with respect to its original counterpart. Nevertheless, it can be preferred only when low noise signals are used for estimation     

Minimum mean squared error impulse noise estimation andcancellation

Impulses are infrequent bursts of high amplitude noise. A wide-band communications or data acquisition receiver has a fast sampling rate, so it can capture many samples of each impulse waveform. The arrival of an impulse can be identified by its distinct waveform and amplitude. The paper models impulse waveforms as a vector subspace of low dimension. Formulas are derived for the minimum mean squared error (MMSE) estimation of the arrival time and amplitudes of impulses, given that a set of vectors that spans the subspace is known. Formulas are also derived for the adaptive MMSE estimation of a set of vectors that spans the subspace. The values of the mean squared error (MSE) of the amplitude estimates are determined. It is shown how the theory can be used to cancel impulse noise. Correlated impulse noise arriving at a reference input can be used to estimate and cancel the primary input impulse noise. The MMSE coefficients for impulse noise cancellation are derived and presented. Simulations are presented that use the equations and methods derived in the paper for modeling and canceling impulse noise measured on copper telephone loops for asymmetric digital subscriber lines (ADSL)     

基于PCNN噪声检测的两级脉冲噪声滤波算法

为有效滤除图像中严重脉冲噪声干扰,提出了一种基于改进型脉冲耦合神经网络(PCNN)噪声检测的两级脉冲噪声滤除算法。该算法首先利用PCNN同步脉冲发放特性区分定位噪声点和信号点位置,其次根据噪声点局部邻域信息对噪声进行第1级自适应滤波,然后再利用具有保护边缘细节特点的多方向信息中值滤波器(MF进行第2级辅助滤波。实验结果表明,该算法在噪声检测中无需设定检测阈值,噪声检测精度较高;在去噪过程中不但有效滤除噪声干扰,而且能很好地保护图像边缘细节等信息,具有较好的主观视觉效果和客观评价指标,比传统MF及其它相关算法有更优的滤波性能,去噪能力强、信噪比高和适应性好,特别是对受严重噪声污染的图像,显示了更大的优越性。     

脉冲噪声环境下改进的顽健循环时延估计算法

研究了脉冲噪声环境下循环平稳信号的时延估计问题,针对脉冲噪声环境中基于传统二阶谱相关函数的时延估计方法性能退化问题,提出了基于分数低阶循环谱的改进顽健算法。相对于传统算法,新算法对脉冲噪声、高斯噪声、干扰信号都具有较好的抑制作用。仿真结果证明了算法的有效性和顽健性。     

改进的变步长变换域最小均方算法

变换域是一种在强相关信号输入时加快自适应算法收敛的方法,但仍然存在收敛速度的要求与稳态失调的要求相矛盾的问题。本文在变换域最小均方误差算法（transform domain LMS, TDLMS）的基础上提出了一种改进的变步长方案,其变步长因子受到误差自相关的控制,消除了不相关的观测噪声的影响。本文分别在平稳和非平稳状态下,对算法的收敛和稳态性能进行理论分析,并给出了最佳的算法参数。仿真设置相同的稳态误差,结果表明本文算法在平稳状态下比固定步长的算法提前1300点收敛,在非平稳状态下提前1400点收敛,且与文献中其它变步长的算法相比收敛速度均有提升。      

A new fuzzy-based decision algorithm for high-density impulse noise removal

This paper proposes a new efficient fuzzy-based decision algorithm (FBDA) for the restoration of images that are corrupted with high density of impulse noises. FBDA is a fuzzy-based switching median filter in which the filtering is applied only to corrupted pixels in the image while the uncorrupted pixels are left unchanged. The proposed algorithm computes the difference measure for each pixel based on the central pixel (corrupted pixel) in a selected window and then calculates the membership value for each pixel based on the highest difference. The algorithm then eliminates those pixels from the window with very high and very low membership values, which might represent the impulse noises. Median filter is then applied to the remaining pixels in the window to get the restored value for the current pixel position. The proposed algorithm produces excellent results compared to conventional method such as standard median filter (SMF) as well as some advanced techniques such as adaptive median filters (AMF), efficient decision-based algorithm (EDBA), improved efficient decision-based algorithm (IDBA) and boundary discriminative noise detection (BDND) switching median filter. The efficiency of the proposed algorithm is evaluated using different standard images. From experimental analysis, it has been found that FBDA produces better results in terms of both quantitative measures such as PSNR, SSIM, IEF and qualitative measures such as Image Quality Index (IQI).     

改进的LMS自适应噪声对消算法的应用

针对工业噪声的特点,将自适应对消算法应用到工业噪声的处理中。根据传统最小均方(Least Mean Square,LMS)自适应算法的缺点,文中通过构造合适的步长因子,引入参数使得算法在提高收敛速度的同时保证较小的稳态误差。放宽算法的约束性条件,以提高步长调整的精度。实验验证,提出的算法与其他算法相比,具有更快的收敛速度、更小的稳态误差以及优良的抗干扰性能。     

冲激噪声下相干信源的求根MUSIC算法

由于在冲激噪声背景下,MUSIC算法对信源波达方向估计将失去韧性,且无法解相干,运算量较大,为此对前后向平滑算法和修正的MUSIC算法进行了改进,提出相干信源波达方向估计新算法。简要分析了线阵模型以及共变,将阵列输出矩阵从二阶原点矩扩展到低阶矩,通过分析共变矩阵,得到基于共变矩阵的空间谱,再将谱峰搜索转变为多项式求根,最后可得相干信源波达方向估计。通过仿真,在冲激噪声或高斯噪声下,改进算法可以对相干信源的波达方向进行正确估计。算法性能分析表明改进算法具有良好的稳健性。     

一种抑制脉冲噪声的边缘检测算法

提出了一种针对脉冲噪声图像的边缘检测算法,算法借鉴了中值滤波的思想,并采用十字型卷积模板计算图像梯度。首先,对参与图像中梯度计算的像素点进行阈值判断,如果是噪声点,该点像素值用3x3窗口中值滤波结果值替代,然后参与梯度计算,如果不是噪声点则直接参与梯度计算;其次对梯度图像进行细化和二值化以提取边缘图像。实验证明,本文算法对脉冲噪声污染图像边缘检测效果良好,较好地抑制了脉冲噪声的影响,而且提取的图像边缘较细,轮廓清晰。和传统的边缘检测算法及基于小波模变换的边缘检测算法相比,算法在抑噪能力上和边缘提取效果上均比较优秀。     

Finite impulse response adaptive self-tuning algorithm

The proposed strategy uses a novel finite impulse response (FIR) and a modified forced response strategy to achieve finite settling time response and does not require either permanent excitation or the solution of a proper Diophantine equation. The method consists of an effective pole removal-zero placement technique     

基于均值梯度的脉冲噪声图像边缘检测算法

为了有效地检测出受脉冲噪声污染图像的边缘,提出了一种基于均值梯度的图像边缘检测算法。算法将检测窗口根据水平和垂直方向分成上、下、左、右4个不同区域,先计算每个区域内非噪声点的平均灰度值,然后利用这些值的差分计算图像梯度,得出梯度图像,最后对梯度图像采用了改进的非极大值抑制方法对梯度图像进行细化并提取边缘。实验结果表明,该算法能够较好地检测出受较高密度脉冲噪声干扰的图像边缘,而且边缘较细,效果明显优于传统Sobel算法,具有较强的实用性。     

An unbiased parametric imaging algorithm for nonuniformly sampled biomedical system parameter estimation

An unbiased algorithm of generalized linear least squares (GLLS) for parameter estimation of nonuniformly sampled biomedical systems is proposed. The basic theory and detailed derivation of the algorithm are given. This algorithm removes the initial values required and computational burden of nonlinear least regression and achieves a comparable estimation quality in terms of the estimates' bias and standard deviation. Therefore, this algorithm is particular useful in image-wide (pixel-by-pixel based) parameter estimation, e.g., to generate parametric images from tracer dynamic studies with positron emission tomography. An example is presented to demonstrate the performance of this new technique. This algorithm is also generally applicable to other continuous system parameter estimation.     

Adaptive threshold nonlinear algorithm for adaptive filters withrobustness against impulse noise

A new adaptation algorithm for adaptive filters is proposed, introducing “adaptive threshold” in the nonlinear correlation function. The adaptive threshold controlled by the long-term average of the error signal power makes the filtering system highly robust against impulse noise, which is demonstrated by the results of simulation and theoretical calculation     

一种基于LMS迭代的环路延时估计算法

数字预失真技术是有效补偿射频功率放大器的非线性方案;然而,前向数据与反馈数据之间的延时估计是数字预失真系统的关键性问题,直接影响着非线性的补偿能力。在此基于滑动窗的相关运算,介绍一种整数倍的环路延时估计算法,同时基于LMS迭代逐个数据输入进行比较,提出了一种自适应小数倍的环路延时估计算法。最后对算法进行仿真验证,结果表明,经过整数倍的环路延时估计后,对于在[-T]s,Ts范围内的残余的小数倍延时,该算法均表现出优异的估计性能。     

基于小波阈值和LMS算法语音降噪算法研究

语音信号降噪是语音信号处理的重要方面,对人们的日常生活有重要影响。常见的降噪算法包括自适应方法及小波变换等。LMS算法是自适应算法的一种,其优点是算法实现简单,复杂度低;其主要缺点是不能同时保证算法的收敛速度和实验精度。小波阈值在处理语音信号方面也有独特的优势,其主要缺点是处理高频信号时存在失真现象。提出将两种方法结合起来,对语音信号进行两次处理,保留了每种算法的优点,同时避免单一方法带来的不利影响。实验结果表明,该方法的效果显著优于单一算法。     

Bayesian regularization and nonnegative deconvolution for room impulse response estimation

This paper proposes Bayesian Regularization And Nonnegative Deconvolution (BRAND) for accurately and robustly estimating acoustic room impulse responses for applications such as time-delay estimation and echo cancellation. Similar to conventional deconvolution methods, BRAND estimates the coefficients of convolutive finite-impulse-response (FIR) filters using least-square optimization. However, BRAND exploits the nonnegative, sparse structure of acoustic room impulse responses with nonnegativity constraints and L/sub 1/-norm sparsity regularization on the filter coefficients. The optimization problem is modeled within the context of a probabilistic Bayesian framework, and expectation-maximization (EM) is used to derive efficient update rules for estimating the optimal regularization parameters. BRAND is demonstrated on two representative examples, subsample time-delay estimation in reverberant environments and acoustic echo cancellation. The results presented in this paper show the advantages of BRAND in high temporal resolution and robustness to ambient noise compared with other conventional techniques.     

Simple and efficient noise estimation algorithm

A simple algorithm is proposed for the estimation of noise in linear active filters. It is also applicable to noise analysis in delta-sigma modulator loops, and related structures. Its usefulness is demonstrated by using it to compare the noise performance of some commonly used filter structures.     

改进的16进制正交幅度调制码的相位估计算法

相干光通信系统中的相位恢复是克服信号收发端和传输过程中产生的相位噪声的必要步骤。为获得低的运算复杂度和良好的相位估计效果,本文对传统的四次方相位估计算法进行了改进,使之具有自适应的数据块长度控制功能。通过理论分析以及Matlab数值仿真方法验证改进的效果。对于16进制正交幅度调制码（16QAM）,改进后的算法在几乎不增...     

An efficient algorithm for the removal of impulse noise from corrupted images

In this letter, we propose an efficient algorithm, which can successfully remove impulse noise from corrupted images while preserving image details. It is efficient, and requires no previous training. The algorithm consists of two steps: impulse noise detection and impulse noise cancellation. Extensive experimental results show that the proposed approach significantly outperforms many other well-known techniques for image noise removal.