Noniterative Automatic Equalization

An equalizer based on the minimum mean-square error (MSE) criterion, but with direct solution of the resulting equations, is described. The direct solution is based on the algorithms devised by Levinson and Trench. With the availability of large-scale integration (LSI) bipolar computing elements, these algorithms are competitive with iterative procedures. A method is considered for the estimation of the required parameters, and for automatic adaption to a changing channel. It is shown that the delay can be adjusted to achieve minimum MSE with respect to that parameter. Simulations undertaken show the robust performance of the algorithm, and that the equalizer performance is not adversely affected by operation in decision-directed mode.     

一种抗飞行器衰落的快速均衡算法

对流层散射通信信道为时变多径信道,当飞行器飞越散射通信链路会导致飞行器衰落。针对飞行器衰落,提出了一种收敛速度快、跟踪能力强、数值稳定性高、复杂度低的快速自适应均衡算法——基于选择更新的累积误差递归最小二乘自适应均衡算法。根据指数加权最小二乘准则,推导出累积误差递归最小二乘算法,依据共轭斜量算法提出抽头系数选择更新准则。均衡算法的复杂度分析和仿真实验表明提出的快速自适应均衡算法不仅复杂度低,而且有效地提高了均衡器克服信道时间衰落的能力。     

自适应均衡算法在信道均衡技术中的应用研究

文中描述了两种非线性均衡器分别为判决反馈均衡器(DFE)和最大似然序列估计(MLSE)均衡器.所用信道模型为加性白高斯噪声信道,在DFE和线性均衡器(LE)中都是使用递归最小二乘(RLS)算法和最小均方(LMS)算法对数据进行分块处理.MLSE均衡器中使用了维特比最佳译码算法.就误比特性能来做以比较,DFE远好于LE,MLSE均衡器又明显优于DFE,并且它能达到几乎最优的性能.     

Channel Equalization Using Adaptive Lattice Algorithms

In this paper, a study of adaptive lattice algorithms as applied to channel equalization is presented. The orthogonalization properties of the lattice algorithms make them appear promising for equalizing channels which exhibit heavy amplitude distortion. Furthermore, unlike the majority of other orthogonalization algorithms, the number of operations per update for the adaptive lattice equalizers is linear with respect to the number of equalizer taps.     

提高旁瓣相消性能的自适应通道均衡技术

在相控阵雷达中,接收通道的幅相特性不一致对自适应旁瓣相消性能的影响很大,本文介绍了一种校正接收通道不一致来提高旁瓣相消性能的自适应均衡算法。仿真结果和性能分析验证了该算法的正确性和有效性。     

Adaptive Equalization Via Fast Quantized-State Methods

In adaptive equalization, there is a tradeoff between convergence rate of the equalizer tap coefficients, the computational speed for each adjustment, the implementation complexity, and algorithm robustness. Parameter update schemes called quantized state (QS) schemes, and fast versions of these schemes termed fast quantized state (FQS) schemes developed within a related context, are here applied to achieve attractive tradeoff options not previously available for adaptive equalization. Three novel simplifications to the QS schemes are introduced and justified by their performance characteristics in adaptive equalization. One simplification is to abandon the likeness to the method of instrumental variables (IV), where the "instrumental variable" is the quantized state vector, and introduce more quantization. Another simplification is to replace asymptotically Toeplitz matrices, or their inverses, by Toeplitz matrices to take advantage of fast schemes for updating QS schemes or taking their inverses.     

适用于水声信道的快速判决反馈盲均衡算法研究

基于对常数模判决反馈算法(CMDFE)进行分析,提出了一种能够快速收敛的判决反馈盲均衡算法。该算法中构造了一种与常数模算法不同的、能够快速收敛的误差函数,并利用该误差函数对前向权进行调整,而反馈权的调整仍使用常数模误差项。前向权和反馈权系数的更新取不同的步长值。最后通过水声信道模型仿真,对这几种算法进行了数值分析研究。结果表明：所提出的算法可有效地实现对多途水声信道的均衡,并且其收敛速度高于常数模算法和常模判决反馈算法。     

卫星通信中快速非线性信道RLS均衡算法

幅度相位联合调制(Amplitude Phase Shift Keying, APSK)信号经过卫星信道时容易引起非线性失真,产生码间串扰。为此,本文将Wiener均衡应用于APSK卫星通信中,并提出了一种快速非线性信道递归最小二乘(Nonlinear Channel Recursive Least Squares, NCRLS)均衡算法,在传统NCRLS算法的基础上,建立了遗忘因子与步长因子的关系,推出了变遗忘因子的非线性信道递归最小二乘(Variable Forgetting Factor Nonlinear Channel Recursive Least Squares, VFFNCRLS)均衡算法,有效地提高了收敛速度。仿真结果表明,VFFNCRLS算法有效地纠正了非线性失真信号的星座扭曲和频谱再生,解决了卫星通信系统中有记忆非线性信道失真的问题,相比传统算法,具有较快的收敛速度和较好的稳定性。      

快速时变信道中几种自适应信道估计算法的比较研究

主要研究了MIMO-OFDM系统中快速时变信道的信道估计问题,将3种自适应信道估计算法: RLS,QRD-RLS,IQRD-RLS,应用于快速时变信道进行比较研究,并对其进行计算机仿真。仿真结果表明,3种算法性能相似,其中,QRD-RLS和IQRD-RLS算法的收敛速度要快于RLS算法,对于实时系统而言,这两种算法的效率更高。因此,QRD-RLS和IQRD-RIS算法更适合于快速时变信道。     

Adaptive Equalization of the Slow Fading Channel

This paper considers equalization of the slow fading channel for a serial data transmission application. Linear and decision-feedback adaptive equalization techniques are contrasted. The error propagation effect in decision-feedback equalizers is analyzed by a Markov process model. The error probability magnification is computed for both fixed and fading channels and for both binary and quaternary phase-shift-keying (PSK) transmission. The results show that the error propagation effect is small and in regions of practical error probabilities the decision-feedback equalizer is superior to its linear counterpart. Parameters of a practical decisionfeedback equalizer are estimated and a performance evaluation is performed. The implicit diversity gain is shown to be significant and the intersymbol interference penalty is found to be less than 1 dB. Because the intersymbol interference penalty is small, more complex nonlinear processors such as the Viterbi algorithm cannot be recommended for this application. Time jitter effects for the equalizer are included in a calculation of average error probability.     

两种修正的自适应通道均衡方法

在常规雷达加装的自适应旁瓣相消(ASLC)系统中,天线至接收机前端馈线部分的相位不一致性通常不可忽略。而现有的自适应通道均衡方法是在接收机前端注入校准信号进行均衡,只能对接收机的幅相不一致性进行校正,对馈线部分的相位失配却无能为力,因此影响了ASLC的对消性能。该文提出了两种修正的自适应通道均衡方法,它们能同时校正接收机与馈线部分的幅相不一致性。理论分析和实验结果验证了这两种方法的有效性。     

一种快速收敛的水声信道分数间隔盲均衡算法

为了加快常数模盲均衡算法(CMA)的收敛速度并降低剩余均方误差,提出了一种快速收敛的分数间隔常数模算法。新算法采用T/2分数间隔均衡器,针对水声通信中常用的BPSK信号,设计了一个新的误差函数,取代常数模算法中的误差函数。最后,采用浅海水声信道进行了仿真,结果表明,新算法性能稳定;收敛速度高于常数模算法和分数间隔常数模算法;剩余均方误差与分数间隔常数模算法处于同一水平。     

宽带卫星信道群时延的自适应均衡校正研究

在宽带卫星通信链路中,由于器件通带特性不理想和行波管放大器的工艺受限等原因使卫星信道的群时延波动较大,导致误比特性能恶化。群时延的补偿算法需要复杂的数学运算,因此工程上一般采用线性均衡对信道群时延特性进行校正。针对群时延失真严重情况下线性均衡效果下降的问题,对比研究了线性均衡、非线性均衡对信道群时延校正的性能,仿真分析了采用恒模算法的线性均衡和采用voherra模型的非线性均衡在群时延失真信道下的性能,得出了两类均衡器在群时延失真信道下的误码率性能曲线,结果表明低阶调制下采用非线性均衡可以较好的消除宽带卫星信道群时延的影响。     

一种水声信道自适应均衡算法研究

由声波传播的多途效应所引起的码间干扰及信号衰落,使得水下数据传输的速率和可靠性都大大降低。针对多途干扰严重的水声信道,将一种基于最小均方误差算法的信道自适应均衡器应用于水声信号处理中。对该均衡器的基本理论及其结构进行了归纳,结合水声多途信道的基本特点,建立了仿真模型,重点分析了步长参数、信噪比等对算法收敛速度和稳态误差等方面性能的影响,为实现高质量的水下信息传递提供技术支持。     

信道估计非迭代算法的进一步研究

本文对文献（１）提出的非迭代信道估计器进行了进一步研究，提出并证明了两个定理，为非迭代法的实际应用提供了理论依据。     

一种直接在中频信号上实现通道均衡的新方法

通道失配严重制约了多通道雷达的性能,诸如旁瓣对消的对消比、数字波束形成的波束旁瓣和单脉冲测角精度等。传统的通道均衡一般是在数字正交下变频之后的基带复信号上通过复数滤波器实现,分析传统均衡滤波器原理后,提出了一种可直接在中频采样后数字正交下变频之前实现通道均衡的新方法,证明了均衡滤波器可以用实系数FIR滤波器完成,仿真验证了该算法的可行性。与传统方法相比,新方法可以降低工程实现的复杂度和运算量,已成功应用于某跟踪雷达中。     

Application of Fast Kalman Estimation to Adaptive Equalization

Very rapid initial convergence of the equalizer tap coefficients is a requirement of many data communication systems which employ adaptive equalizers to minimize intersymbol interference. As shown in recent papers by Godard, and by Gitlin and Magee, a recursive least squares estimation algorithm, which is a special case of the Kalman estimation algorithm, is applicable to the estimation of the optimal (minimum MSE) set of tap coefficients. It was furthermore shown to yield much faster equalizer convergence than that achieved by the simple estimated gradient algorithm, especially for severely distorted channels. We show how certain "fast recursive estimation" techniques, originally introduced by Morf and Ljung, can be adapted to the equalizer adjustment problem, resulting in the same fast convergence as the conventional Kalman implementation, but with far fewer operations per iteration (proportional to the number of equalizer taps, rather than the square of the number of equalizer taps). These fast algorithms, applicable to both linear and decision feedback equalizers, exploit a certain shift-invariance property of successive equalizer contents. The rapid convergence properties of the "fast Kalman" adaptation algorithm are confirmed by simulation.     

盲均衡的一种快速算法

本文提出了盲均衡的一种快速算法.该方法与通常的盲均衡算法有着本质的区别,它不需要使用代价函数,也就不需要一个最小化代价函数的算法,从而避免了一些盲均衡器不能全局收敛的问题.本算法基于信道输入信号的一些先验的统计知识,并充分运用了信道输入序列与信道输出观测序列之间的关系.新算法主要的运算为比较运算,只用到了少量的乘、加运算,所以计算量远小于通常的盲均衡器算法.     

一种快速时变自适应信道估计盲方法

提出了一种新的信道估计最小均方(LMS)盲方法--时变步长理想加权判决信道估计盲方法.该方法通过对常规LMS算法进行改进,实现自适应跟踪信道特征变化;同时利用接收机判决误差信息函数作为权系数的理想加权判决,从而解决了常规LMS盲方法收敛速度慢、信道估计性能不高的缺点.仿真证明,对于不同的多普勒频移、时延扩展以及时间,该方法均表现出了比常规LMS方法更优的性能.同时,该方法还可用于航天、雷达等领域估计其他特征参数.     

A Microprocessor-Oriented Algorithm for Adaptive Equalization

Adaptive equalization based on a transversal filter is extremely important for reliable data transmission over the voiceband telephone lines. Unfortunately, implementation of rapidly converging and small residual distortion algorithms on the standard MOS microprocessors or standard-cell MOS VLSI chips is difficult because of their mathematical complexity. In this paper, an adaptive equalization algorithm is proposed which is suitable for cost-effective implementation on MOS microprocessors and standard-cell MOS VLSI chips. It is based on the same principle of delayed decision that was used successfully in delta modulation. It involves no multiplications or analog-to-digital conversion, but is relatively fast and provides a relatively low residual distortion. The proposed algorithm is simulated in both noise-free and noisy environments and is compared to two well-known classical algorithms.