2008Noise Robust Multichannel Frequency-Domain LMS Algorithms for Blind Channel Identification
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低精度ADC 下的大规模MIMO-OFDM 信道估计算法戈立军,朱德宝(天津工业大学电子与信息工程学院,天津300387)Channel estimation algorithm for massive MIMO-OFDM systems withlow-precision ADCsGE Li-jun ,ZHU De-bao(School of Electronics and Information Engineering ,Tiangong University ,Tianjin 300387,China )Abstract :A channel estimation algorithm based on quantized compressive sensing is proposed for massive multiple inputmultiple output-orthogonal frequency division multiplexing渊MIMO-OFDM冤systems with low-precision analog-to-digital converters (ADCs )袁which is the block sparsity multi-bit iterative hard thresholding 渊B-MIHT冤algo鄄rithm.The B-MIHT algorithm exploits the block sparsity characteristics of massive MIMO-OFDM system chan鄄nels袁and combines with the multi-bit iterative hard thresholding algorithm by constructing the equivalent blocksparse channel matrices.The algorithm estimates the channel information of massive MIMO-OFDM systems withlow-precision ADCs based on training sequences袁the simulation is performed on MATLAB platform.The results show that B-MIHT algorithm can accurately recover the channel information of massive MIMO-OFDM systems with low -precision ADCs and has good channel estimation performance under the condition that the system quantization accuracy is 5bits.When the signal to noise ratio is 30dB袁the bit error rate渊BER冤of B-MIHT algo鄄rithm is 5.45伊10-3and the normalized mean square error渊NMSE冤is 1.73伊10-3.The channel estimation perfor鄄mance loss of B-MIHT algorithm is relatively small when the number of channel paths increases.Key words :massive multiple input multiple output-orthogonal frequency division multiplexing (MIMO-OFDM );low-precision analog-to-digital converter (ADC );channel estimation ;quantized compressive sensing ;blocksparsity characteristic摘要:针对低精度模数转换器(ADC )下的大规模多输入多输出正交频分复用(MIMO-OFDM )系统,提出一种基于量化压缩感知的信道估计算法———块稀疏多比特迭代硬阈值(B-MIHT )算法。
第34卷第2期电子与信息学报Vol.34No.2 2012年2月 Journal of Electronics & Information Technology Feb. 2012用于宽带频谱感知的全盲亚奈奎斯特采样方法盖建新①②付平*①乔家庆①孟升卫①③①(哈尔滨工业大学自动化测试与控制系哈尔滨 150080)②(哈尔滨理工大学测控技术与仪器黑龙江省高校重点实验室哈尔滨 150080)③(中国科学院电子学研究所北京 100190)摘要:亚奈奎斯特采样方法是缓解宽带频谱感知技术中采样率过高压力的有效途径。
该文针对现有亚奈奎斯特采样方法所需测量矩阵维数过大且重构阶段需要确切稀疏度的问题,提出了将测量矩阵较小的调制宽带转换器(MWC)应用于宽带频谱感知的方法。
在重新定义频谱稀疏信号模型的基础上,提出了一个改进的盲谱重构充分条件,消除了构建MWC系统对最大频带宽度的依赖;在重构阶段,将稀疏度自适应匹配追踪(SAMP)算法引入到多测量向量(MMV)问题的求解中。
最终实现了既不需要预知最大频带宽度也不需要确切频带数量的全盲低速采样,实验结果验证了该方法的有效性。
关键词:宽带频谱感知;亚奈奎斯特采样;多测量向量;稀疏度自适应匹配追踪中图分类号:TN911.72 文献标识码:A 文章编号:1009-5896(2012)02-0361-07 DOI: 10.3724/SP.J.1146.2011.00314A Full-blind Sub-Nyquist Sampling Methodfor Wideband Spectrum SensingGai Jian-xin①②Fu Ping① Qiao Jia-qing① Meng Sheng-wei①③①(Department of Automatic Test and Control, Harbin Institute of Technology, Harbin 150080, China)②(The Higher Educational Key Laboratory for Measuring & Control Technology and Instrumentations of Heilongjiang Province,Harbin University of Science and Technology, Harbin 150080, China)③(Institute of Electronics, Chinese Academy of Sciences, Beijing 100190, China)Abstract: Sub-Nyquist sampling is an effective approach to mitigate the high sampling rate pressure for wideband spectrum sensing. The existing sub-Nyquist sampling method requires excessive large measurement matrix and exact sparsity level in recovery phase. Considering this problem, a method of applying Modulated Wideband Converter (MWC) with small measurement matrix to wideband spectrum sensing is proposed. An improved sufficient condition for spectrum-blind recovery based on the redefinition of spectrum sparse signal model is presented, which breaks the dependence on the maximum width of bands for MWC construction. In recovery phase, the Sparsity Adaptive Matching Pursuit (SAMP) algorithm is introduced to Multiple Measurement Vector (MMV) problem. As a result, a full-blind low rate sampling method requiring neither the maximum width nor the exact number of bands is implemented. The experimental results verify the effectiveness of the proposed method.Key words: Wideband spectrum sensing; Sub-Nyquist sampling; Multiple Measurement Vectors (MMV); Sparsity Adaptive Matching Pursuit (SAMP)1引言认知无线电通过感知周围频谱环境自主发现“频谱空穴”并对其进行有效利用,在解决无线通信中频谱资源紧张、频谱利用率低等问题上表现出巨大的优势。
2008 年 12月 JOURNALOF CIRCUITS AND SYSTEMS December , 2008 文章编号:1007-0249 (2008) 06-0034-06多径衰落信道下基于子空间的RLS 盲多用户检测*兰元荣1, 王东昱2, 武梦龙2, 杨大成1(1. 北京邮电大学 93#,北京邮电大学 无线通信中心,北京 100876;2. 北方工业大学 信息工程学院,北京 100041)ᐢገǖ本文在RLS 盲检测算法的基础上,利用子空间的概念,构建了基于子空间的RLS 多用户盲检测算法,在仅仅需要知道目标用户的特征序列和定时的条件下,自适应地估计检测向量,通过理论分析表明,改进的检测算法在运算复杂度上低于满秩RLS 算法[7]。
仿真结果表明,改进的检测算法收敛性能优于满秩RLS 算法,同时在特征序列畸变条件下表现出健壮性也远优于满秩RLS 检测算法。
ਈࠤǖ盲多用户检测;DS/CDMA ;MAI ;RLS 滤波算法;收敛性分析 ᒦᅄॊಢǖTN92ᆪማܪဤ൩ǖA1 引言MAI 是CDMA 系统最主要的干扰,因此消除MAI 对系统性能的影响一直是研究的一个热点,为能够成功地消除MAI 并正确地检测出目标用户的信息比特,在接收端应该知以下的一个或者几个量:1)目标用户的扩频序列;2)其它干扰用户的扩频序列;3)目标用户的传输时延;4)其它干扰用户的传输时延;5)其它干扰用户相对于目标用户接收信号的幅度;6)每一个用户的训练序列。
最佳的多用户接收机由Verdu [1]提出,在需要知道1)~5)的情况下,其计算复杂度随着用户数目的呈指数倍增长,因此不可能在实际中运用。
解相关多用户检测[2],最小均方误差估计(MMSE )以及等价的MOE [3,4]和最佳检测器[1]相比较,复杂度得到了降低。
[5]中的解相关多用户检测器和[6]自适应MMSE 检测器需要为每一个用户进行序列训练。
相比较而言,自适应盲多用户检测则和传统接收机一样,只需要知道1)和3),近年来,有关盲多用户的检测的研究见诸于文献[3,7~14]。
一种用于自适应噪声抵消的变步长LMS算法
李善姬
【期刊名称】《电讯技术》
【年(卷),期】2010(50)11
【摘要】为了提高LMS自适应滤波算法的性能,在分析已有变步长算法的基础上进行了一些改进.改进算法用误差信号的自相关来调节步长以实现对不相关噪声的更好抑制,且采用先固定后变化的方法控制步长,兼顾了暂态和稳态性能.利用改进算法进行了自适应噪声抵消的仿真实验,结果表明,基于改进变步长LMS算法的自适应噪声抵消器能有效抵制噪声干扰,对含噪信号具有良好的消噪能力.
【总页数】4页(P30-33)
【作者】李善姬
【作者单位】延边大学,工学院,吉林,延吉,133002
【正文语种】中文
【中图分类】TN911.7
【相关文献】
1.改进变步长LMS算法及在自适应噪声抵消中的应用 [J], 沈磊;姚善化
2.一种应用于自适应降噪的变步长LMS算法 [J], 邵成华;姚善化
3.一种应用于自适应降噪的变步长LMS算法 [J], 杜秀群;冯西安;杜伟
4.一种用于信号估计的改进变步长LMS算法 [J], 刘牮;李彧
5.一种适用于水声环境的变步长LMS算法 [J], 罗斌;张海生;王晓林
因版权原因,仅展示原文概要,查看原文内容请购买。
LMS自适应滤波算法1960年Widrow和Hoff提出最小均方误差算法(LMS),LMS算法是随机梯度算法中的一员。
使用“随机梯度”一词是为了将LMS算法与最速下降法区别开来。
该算法在随机输入维纳滤波器递归计算中使用确定性梯度。
LMS算法的一个显著特点是它的简单性。
此外,它不需要计算有关的相关函数,也不需要矩阵求逆运算。
由于其具有的简单性、鲁棒性和易于实现的性能,在很多领域得到了广泛的应用。
1LMS算法简介LMS算法是线性自适应滤波算法,一般来说包含两个基本过程:(1)滤波过程:计算线性滤波器输出对输入信号的响应,通过比较输出与期望响应产生估计误差。
(2)自适应过程:根据估计误差自动调整滤波器参数。
如图1-1所示,用表示n时刻输入信号矢量,用表示n时刻N阶自适应滤波器的权重系数,表示期望信号,表示误差信号,是主端输入干扰信号,u是步长因子。
则基本的LMS算法可以表示为(1)(2)图1-1 自适应滤波原理框图由上式可以看出LMS算法实现起来确实很简单,一步估计误差(1),和一步跟新权向量(2)。
2迭代步长u的作用2.1 理论分析尽管LMS算法实现起来较为简单,但是精确分析LMS的收敛过程和性能却是非常困难的。
最早做LMS收敛性能分析的是Widrow等人,他们从精确的梯度下降法出发,研究权矢量误差的均值收敛特性。
最终得到代价函数的收敛公式:′(3)式(3)揭示出LMS算法代价函数的收敛过程表现为一簇指数衰减曲线之和的形式,每条指数曲线对应于旋转后的权误差矢量的每个分量,而他们的衰减速度,对应于输入自相关矩阵的每个特征值,第i条指数曲线的时间常数表示为τ小特征值对应大时间常数,即衰减速度慢的曲线。
而大特征值对应收敛速度快的曲线,但是如果特征值过大以至于则导致算法发散。
从上式可以明显看出迭代步长u在LMS算法中会影响算法收敛的速度,增大u可以加快算法的收敛速度,但是要保证算法收敛。
最大步长边界:稳态误差时衡量LMS算法的另一个重要指标,稳定的LMS算法在n时刻所产生的均方误差,其最终值∞是一个常数。
频域lms算法范文频域Least Mean Square (LMS)算法是一种基于自适应滤波的算法,广泛应用于信号处理领域。
其主要思想是通过不断地调整滤波器的系数,使滤波器的输出尽可能接近期望输出。
频域LMS算法在频域上运算,能够对频域信息进行处理,因此具有一定的优势。
频域LMS算法的核心是通过最小化均方误差的方式来调整滤波器系数。
假设有一个期望输出序列d(n)和一个输入序列x(n),我们的目的是找到一个滤波器的系数向量W(n),使得滤波器的输出y(n)与期望输出d(n)的误差e(n)最小。
通过不断地调整滤波器的系数,使误差e(n)达到最小,从而实现信号的滤波。
频域LMS算法的步骤如下:1.将输入序列x(n)和期望输出序列d(n)进行傅里叶变换,得到频域上的输入信号X(k)和期望输出信号D(k)。
2.初始化滤波器的系数向量W(n)为零向量。
3.对于每一个输入样本,初始化预测输出y(n)为滤波器的输出信号的傅里叶反变换。
4.计算误差信号e(n)=D(k)-Y(k)的傅里叶变换,其中Y(k)为滤波器的输出信号的傅里叶变换。
5.更新滤波器的系数向量W(k)=W(k-1)+μ*X'(k)*e'(k)/(X(k)*X'(k)+λ),其中μ为步长因子,X'(k)和e'(k)为X(k)和e(k)的共轭。
6.重复步骤3-5,直至满足收敛条件。
频域LMS算法的优点是能够处理频域信息,对频域信号具有较好的处理能力。
此外,频域LMS算法具有较快的收敛速度和较低的计算复杂度,适用于实时信号处理。
然而,频域LMS算法也存在一些缺点。
首先,频域LMS算法需要进行傅里叶变换和反变换,因此需要较大的计算开销。
其次,频域LMS算法对信号的稳态行为要求较高,对非稳态信号处理效果较差。
此外,频域LMS算法对噪声的统计特性有一定的要求,对于非高斯白噪声的处理效果较差。
总之,频域LMS算法是一种自适应滤波算法,能够对频域上的信号进行处理。
改进的频域窄带干扰抑制方法李平博;王璐;严玉国【摘要】针对频域窄带干扰抑制方法的不足,分别提出改进的自适应多门限干扰检测算法和广义延拓逼近算法在干扰抑制中的应用.改进算法使用去除干扰频点之后的频谱进行检测干扰,检测效果有了明显提高;有约束条件的一元函数延拓逼近算法,结构简单,通过延拓逼近思想,减小了对干扰频点处理时信号时域波形的失真.Matlab仿真通过误码率对比验证了算法性能,从而证明改进方法具有更好的干扰检测性能和更优的干扰处理能力.【期刊名称】《空军工程大学学报(自然科学版)》【年(卷),期】2015(016)002【总页数】4页(P78-81)【关键词】快速傅里叶变换;窄带干扰;窄带高斯模型;延拓逼近【作者】李平博;王璐;严玉国【作者单位】空军工程大学信息与导航学院,西安,710077;铁道警察学院,郑州,450053;空军工程大学信息与导航学院,西安,710077【正文语种】中文【中图分类】TN914.42直扩通信由于能有效地抑制窄带干扰而得到了广泛的应用和研究[1-2]。
然而扩频增益是有限的,在强窄带干扰场合必须通过其他方法来进一步提高扩频通信的抗干扰能力。
其中,变换域滤波器技术是抗窄带干扰的一类最有用的方法[3-4]。
根据信号变换方式、干扰检测和陷波算法的不同,基于变换域的干扰抑制方式有很多,其中,基于傅里叶变换的变换域抗干扰技术由于其实现简单,硬件资源占用少和处理速度快等优点在工程实践中得到了广泛应用。
基于傅里叶变换的窄带干扰抑制方法一般有2个弱点[4]:一是难以快速、准确地检测或估计干扰;二是消除干扰的同时容易损伤信号,尤其是对于快时变的复杂干扰。
针对问题,本文提出了改进的自适应多门限干扰检测算法和广义延拓逼近算法在干扰抑制中的应用。
传统的频域干扰抑制算法基本思路见图1。
根据干扰检测机理的不同和频谱抑制方式的区别,频域干扰抑制算法有中值滤波法、权值泄露法、K谱线法以及门限检测法等。
频域lms算法频域LMS算法是一种常用的自适应滤波算法,主要用于信号处理和系统辨识等领域。
本文将介绍频域LMS算法的原理、应用以及优缺点。
一、频域LMS算法原理频域LMS算法是基于最小均方(Least Mean Square,LMS)准则的自适应滤波算法。
其主要思想是通过最小化误差信号的均方差,来不断调整滤波器的系数,从而实现滤波器的自适应更新。
具体来说,频域LMS算法将输入信号和滤波器系数都转化到频域进行处理。
首先,将输入信号和滤波器系数都进行傅里叶变换,得到它们的频域表示。
然后,根据LMS准则,通过计算误差信号的均方差梯度来更新滤波器系数。
最后,将更新后的频域滤波器系数进行反傅里叶变换,得到时域滤波器系数,从而实现滤波器的更新。
二、频域LMS算法应用频域LMS算法在信号处理和系统辨识等领域有着广泛的应用。
以下是几个常见应用场景:1. 自适应滤波:频域LMS算法可以用于自适应滤波,通过不断调整滤波器系数,从输入信号中提取出所需的信息,抑制不需要的噪声和干扰。
这在语音增强、图像去噪等领域有着重要的应用。
2. 信道均衡:在通信系统中,信道的非理想性会引入干扰和失真,影响系统性能。
频域LMS算法可以用于信道均衡,通过自适应滤波来抵消信道引入的失真,从而提高系统的传输性能。
3. 系统辨识:频域LMS算法可以用于系统辨识,通过分析输入信号和输出信号之间的关系,从中提取出系统的特征和参数。
这在控制系统设计和模型建立中起到了重要作用。
三、频域LMS算法优缺点频域LMS算法具有以下优点:1. 计算效率高:由于频域LMS算法将信号和滤波器系数都转化到频域进行处理,可以利用快速傅里叶变换等高效算法,提高计算效率。
2. 收敛速度快:频域LMS算法可以通过选择合适的步长参数和初始化滤波器系数,提高算法的收敛速度。
3. 适用性广:频域LMS算法可以应用于各种信号处理和系统辨识问题,具有较好的通用性。
然而,频域LMS算法也存在一些缺点:1. 算法复杂度高:频域LMS算法需要进行频域转换和反转换操作,增加了算法的复杂度和计算开销。
IEEE SIGNAL PROCESSING LETTERS, VOL. 15, 2008305Noise Robust Multichannel Frequency-Domain LMS Algorithms for Blind Channel IdentificationMohammad Ariful Haque and Md Kamrul Hasan, Senior Member, IEEEAbstract—A number of multichannel least mean square (LMS)-type algorithms have been proposed in the literature to identify single-input multi-output finite impulse response channels. All of these algorithms share the common characteristic of good initial convergence followed by a rapid misconvergence in the presence of noise. This misconvergence characteristic is due to the nonuniform spectral attenuation of the estimated channel coefficients as reported in some research results. In this letter, we formulate a novel cost function that inherently oppose such spectral attenuation resulting from the noisy update vector. We show analytically that the gradient of the proposed penalty term enforces uniform distribution of the estimated channel spectral energy over the entire frequency band and thus contribute to ameliorating the misconvergence of these multichannel algorithms in the presence of noise. The robustness of the proposed algorithm is verified using numerical examples for different channels in a wide range of signal-to-noise ratios. Index Terms—Blind channel identification, multichannel least mean square (LMS) algorithm, noise robustness.I. INTRODUCTION LIND channel identification (BCI) is a signal processing technology aimed to identify the unknown channel impulse responses solely from the observations often corrupted by noise. Both single- and multichannel identification schemes are reported in the literature by many researchers. Multichannel identification schemes, however, are increasingly becoming popular due to their suitability in removing the unknown channel effects more effectively than their single-channel counterparts. Among the various multichannel BCI schemes reported so far, e.g., least-squares approach [1], subspace method [2], maximum-likelihood method [3], the normalized multichannel frequency-domain LMS (NMCFLMS) [4], and the variable step-size multichannel frequency-domain LMS (VSS-MCFLMS) [5], algorithms are attractive as they are computationally efficient and effective for BCI. It is shown in [6] that the VSS-MCFLMS algorithm is more robust as compared to the NMCFLMS algorithm in the presence of noise. However, the NMCFLMS algorithm has higher speed of convergence in acoustic multichannel systems followed by a rapid misconvergence. Therefore, if the robustness of the NMCFLMS algorithm could be ensured, it would be an attractive algorithm for acoustic channel estimation. A constrained NMCFLMS algorithm has been described in [7] for improving its robustness to additive noise. However, theBimplementation of the algorithm assumes that the positions and amplitudes of these dominant components, e.g., direct-path coefficients of the acoustic impulse responses, are known, which restricts its use in practical situations. Moreover, the algorithm is not suitable for random coefficients systems where there are no such dominant components. Recently, it has been reported in [8] and [9] that the misconverged estimates of the channel impulse responses have narrowband characteristics in the frequency-domain. An energy constrained NMCFLMS algorithm for blind identification of acoustic channels is proposed in [9] which is based on achieving the spectral energy balance between the low- and high-frequency subbands of the estimated channel. However, the technique is not generalized to include the total misconvergence scenario. For example, if the narrowband phenomenon appears both in the low- and high-frequency bands, the constraint does not work. Moreover, we have observed that the technique is not useful to stop the misconvergence when the channel coefficients are white random sequence. In this letter, we propose a robust technique that can stop the misconvergence of the NMCFLMS and VSS-MCFLMS algorithms in blind identification of both acoustic and random channels. We formulate a mathematically tractable penalty function to enforce that the spectral energy is approximately uniformly distributed over all the frequencies in the estimated channel response. Thus, the gradient of this penalty function works against the nonuniform spectral attenuation effect of the original noisy gradient and hence ensures robustness of the adaptive blind multichannel algorithms. This penalty term can be directly calculated from the estimated impulse responses, and therefore, it is implementable without a priori information of the channel and observation noise power. II. PROBLEM FORMULATION The input–output relationship of a single-input multipleoutput (SIMO) finite impulse response (FIR) channel is given by (1) (2) where is the number of sensors, is the length of the im, , , , and denote, respecpulse response, tively, the common source signal, th channel output, th channel output corrupted by background noise, observation noise, and impulse response of the source to the th sensor. Using vector notation, (1) can be written as (3) where, “ ” denotes transpose is and the impulseManuscript received October 24, 2007; revised December 15, 2007. The associate editor coordinating the review of this manuscript and approving it for publication was Prof. Yue (Joseph) Wang. The authors are with the Department of Electrical and Electronic Engineering, Bangladesh University of Engineering and Technology, Dhaka, Bangladesh (e-mail: arifulhoque@eee.buet.ac.bd; khasan@eee.buet.ac.bd). Color versions of one or more of the figures in this paper are available online at . Digital Object Identifier 10.1109/LSP.2008.9178031070-9908/$25.00 © 2008 IEEE306IEEE SIGNAL PROCESSING LETTERS, VOL. 15, 2008response vector of the estimates observations the data length.,th channel and . A BCI algorithm solely from the noisy , where denotesnorm in the time-domain. Imposing this constraint in the frequency-domain, we have the update equation as (9) It is observed that both the NMCFLMS and VSS-MCFLMS algorithms give good initial estimate of the channels followed by rapid divergence from this better estimate in the presence of additive noise. This misconvergence is associated with the nonuniform spectral attenuation of the estimated channel impulse response [8], [9]. They proposed a modified cost function , where and are the original and penalty cost functions, respectively, which are cou. was formupled through the Lagrange multiplier, , where denotes the frelated as quency band over where spectral energy has been concentrated indicates due to misconvergence. Though minimization of that energy in a particular band is minimized, but this does not mean that energy will be uniformly distributed over the entire is minimized, but the frequency band. It is not unlikely that shape of the spectrum is still narrow band. Therefore, we propose a novel penalty function that ensures spectral flatness of the estimated channel coefficients in the presence of additive noise. The penalty function that can ameliorate the misconvergence of the MCFLMS-type algorithms is defined in this work as maximize subject to (11) where (11) is ensured by the unit norm constraint imposed on the update equation. Now substituting (11) into (10), we obtain (10)III. ROBUST MCFLMS-TYPE ALGORITHMS We first briefly review the multichannel frequency-domain LMS-type algorithms for blind channel identification. The cost function in the frequency-domain is defined as (4) is the frequency-domain block error signal bewhere tween the th and th channels and the superscript “ ” denotes Hermitian transpose. The update equation of the NMCFLMS algorithm [4] can be expressed as(5) where algorithm and is the step-size for the NMCFLMS , and . Here, “ ” indicates estimate, is the frame index, and denotes the DFT matrix. is given by The frequency-domain error function (6) is the DFT of the th frame data The diagonal matrix block for the th channel, i.e., see the equation at the bottom of the page. Again, the update equation of the VSS-MCFLMS algorithm [5] can be expressed as (7) is the gradient of the cost function and is where the step-size. Here, the step-size is adapted so that the distance and is minimum at each iteration, and it is between expressed as (8)(12) Differentiating (12) with respect to , we get(13) The penalty function is maximized when (13), we see that it is only possible when . From(14) is the norm and is a small positive real where number used to prevent singularity. To avoid a trivial estimate with all zero elements, at each step, the estimated filter coefficient vector is constrained to have unit If any other term in (13) is equated to zero, we obtain , which denotes the minimum value of the cost function. As we intend to maximize the penalty function, this is a trivial solution. simultaneous In a similar approach, we can obtainHAQUE AND HASAN: NOISE ROBUST MULTICHANNEL FREQUENCY-DOMAIN LMS ALGORITHMS307linear equations of the same form as (14) for each value of . Adding all such equations, we get, is estimated such that the total The coupling factor, in the steady-state condigradient becomes zero , and premultiplying tion. This gives both sides by , we can obtain as (19) Similarly, the update equation for the robust NMCFLMS (RNMCFLMS) algorithm can be written as(15) Subtracting (15) from (14), we obtain the condition for penalty . Thus, the function maximization as penalty function will be maximum when the estimated channel coefficients have uniform magnitude spectra in the frequencydomain. Therefore, to combat nonuniform spectral attenuation problem in the misconvergence phase, spectral flatness can be , attached as a constraint with the original cost function, , using the Lagrange multiplier. Unvia the penalty term, like the modified cost function proposed in [9], the total regularized cost function to be minimized can be defined as . The negative sign before enis minimized, is maximized. The sures that while adaptive update rule for this constrained minimization can be readily obtained as(20) where is estimated similar to (19) but using the NMCFLMS algorithm update parameters. The extra computational cost required to implement the proposed penalty term is not significant. For example, the total number of multiplications and divisions required by the NMCFLMS algorithm is per iteration, whereas the increase in the com. putational cost due to the added penalty term is only IV. SIMULATION RESULTS In this section, we investigate the performance of the proposed RNMCFLMS and RVSS-MCFLMS algorithms in (20) and (16) for both random and acoustic multichannel systems. We also compare the performance of the proposed method with that of the original NMCFLMS, VSS-MCFLMS, and energy constrained NMCFLMS (CNMCFLMS) [9] algorithms in different noisy environments to show the robustness of our method. In all cases, the step-size parameter, , for the NMCFLMS and RNMCFLMS algorithms was fixed to 0.5, unless otherwise stated. The performance index used for measurement of improvement and comparison is the normalized projection misalignment (NPM) as mentioned in [9]. A. Random Multichannel System We now present blind identification results for a channel random coefficient impulse response system. The impulse responses were generated using the “randn” function of MATLAB. The length of each channel impulse response is . The source signal was Gaussian white noise. Fig. 1 shows the results of channel estimation for all the algorithms at dB. In terms of final misalignment error, the RNMSNR CFLMS algorithm shows the best performance while the original NMCFLMS algorithm without the penalty term demonstrates the poorest performance. The CNMCFLMS algorithm though performs slightly better than the NMCFLMS and is not robust for random coefficient channel as mentioned before in Section I. The results in Fig. 1 also demonstrate that the VSS-MCFLMS algorithm is more robust to noise as compared to the NMCFLMS algorithm in the blind identification of random channels. As it is not showing misconvergence at this SNR value for random coefficient channel estimation, the RVSS-MCFLMS does not bring further improvement in this case.(16) The beauty of the proposed penalty function is that its gradient remains almost inactive as compared to the original signal gradient in the initial phase of iterations. This phenomena stems from the fact that the true channel vector, whether it is acoustic or random, is spectrally wide band. Thus, the original cost function is expected to be better minimized with a wide band estimate of the channel. This leads to almost negligible gradient of the penalty term, thereby making no noticeable effect on the update equation. However, when the misconvergence starts because of nonuniform spectral attenuation in the estimate, the initially dormant gradient of the penalty term becomes active, enforces spectral flatness, and eventually eradicates misconvergence. In order to simplify the expression of the penalty gradient, we take natural a logarithm on both sides of (10). This does not relax the functionality of the penalty term. Therefore, we can rewrite the penalty cost function as (17) Now the penalty function gradient is obtained aswherecan be expressed aswhere Therefore, we can writeand as.(18) where , is a diagonal matrix with diagonal elements .308IEEE SIGNAL PROCESSING LETTERS, VOL. 15, 2008Fig. 1. NPM profile for M = 5 channels L = 32 random coefficients SIMO channel identification at SNR = 10 dB.Fig. 4. NPM profile for a M = 5 channel L = 128 coefficients acoustic channel identification with speech input at SNR = 15 dB.file of the estimated channel using speech input at SNR 15 dB in Fig. 4. In case of the NMCFLMS and VSS-MCFLMS algorithms, we see good initial convergence. With increased iterations, the NMCFLMS completely misconverges. As stated earlier, the VSS-MCFLMS is more robust than the NMCFLMS and shows a slow misconverging trend. To the contrary, the proposed RNMCFLMS and RVSS-MCFLMS algorithms show no sign of misconvergence with noise.Fig. 2. NPM profile for a M = 5 channel L = 256 coefficients acoustic channel identification with white Gaussian input at SNR = 10 dB.V. CONCLUSION A noise robust technique for adaptive blind channel estimation using multichannel frequency-domain LMS-type algorithms has been presented in the letter. We have derived a penalty function whose gradient effectively works against the nonuniform spectral attenuation phenomenon in the estimated channel impulse responses caused by the noisy gradient of the conventional cost function. Computer simulation tests have demonstrated that the proposed technique with complementary cost functions gives reasonably good final misalignment error and improved robustness to additive noise without sacrificing the speed of convergence. REFERENCES[1] G. Xu, H. Liu, L. Tong, and T. Kailath, “A least-squares approach to blind channel identification,” IEEE Trans. Signal Process., vol. 43, no. 12, pp. 2982–2993, Dec. 1995. [2] S. Gannot and M. Moonen, “Subspace methods for multimircophone speech dereverberation,” EURASIP J. Appl. Signal Process., vol. 2003, no. 11, pp. 1074–1090, 2003. [3] Y. Hua, “Fast maximum likelihood for blind identification of multiple FIR channels,” IEEE Trans. Signal Process., vol. 44, no. 3, pp. 661–672, Mar. 1996. [4] Y. Huang and J. Benesty, “A class of frequency-domain adaptive approaches to blind multichannel identification,” IEEE Trans. Signal Process., vol. 51, no. 1, pp. 11–24, Jan. 2003. [5] M. A. Haque and M. K. Hasan, “Variable step size frequency domain multichannel LMS algorithm for blind channel identification with noise,” in Proc. Communication Systems, Networks and Digital Signal Processing, 2006. [6] M. A. Haque and M. K. Hasan, “Performance comparison of the blind multi channel frequency domain normalized LMS and variable step-size LMS with noise,” in Proc. European Signal Processing Conf., 2007. [7] M. K. Hasan, J. Benesty, P. A. Naylor, and D. B. Ward, “Improving robustness of blind adaptive multichannel identification algorithms using constraints,” in Proc. Eur. Signal Processing Conf., 2005. [8] M. K. Hasan and P. A. Naylor, “Effect of noise on blind adaptive multichannel identification algorithms: Robustness issue,” in Proc. Eur. Signal Processing Conf., 2006. [9] M. A. Haque, M. Bashar, P. Naylor, K. Hirose, and M. K. Hasan, “Energy constrained frequency-domain normalized LMS algorithm for blind channel identification,” in J. Signal, Image and Video Process.. London, U.K.: Springer, Apr. 2007, pp. 203–213.Fig. 3. NPM versus SNR profile for M = 5 channel L = 128 coefficients acoustic channel identification with white Gaussian input.B. Acoustic Multichannel System The dimension of the room was taken to be (5 4 3) m. microphones with uniform A linear array consisting of along the -axis was used in the exseparation of periment. The first microphone and source were positioned at (1.0,1.5,1.6) m and (2.0,1.2,1.6) m, respectively. The impulse responses were generated using the well-known image model . The sampling frequency for reverberation time was 8 kHz. dB The results of acoustic channel estimation at SNR is shown in Fig. 2. It can be observed that the penalty term ensures robustness of both the algorithms without sacrificing the speed of convergence for the reason explained after (16). It can also be seen that the step-size parameter, , acts as a trade-off between the convergence speed and final misalignment error for the proposed RNMCFLMS algorithm. Now, to observe the result of channel estimation at different SNRs, we present the final NPM versus SNR profile in Fig. 3. As can be seen, the proposed algorithms give significantly lower final misalignment error as compared to the original ones. We now present the NPM pro-。