Combined RLS-LMS initialization for per tone equalizers in DMT-receivers

In discrete multitone receivers, the classical equalizer structure consists of a (real) time domain equalizer (TEQ) combined with complex one-tap frequency domain equalizers. An alternative receiver is based on a per tone equalization (PTEQ), which optimizes the signal-to-noise ratio (SNR) on each tone separately and, hence, the total bitrate. In this paper, a new initialization scheme for the PTEQ is introduced, based on a combination of least mean squares (LMS) and recursive least squares (RLS) adaptive filtering. It is shown that the proposed method has only slightly slower convergence than full square-root RLS (SR-RLS) while complexity as well as memory cost are reduced considerably. Hence, in terms of complexity and convergence speed, the proposed algorithm is in between LMS and RLS.     

IQ-Imbalance Compensation for OFDM in the Presence of IBI and Carrier-Frequency Offset

In this paper, we propose a frequency-domain equalization technique for orthogonal frequency division multiplexing (OFDM) transmission over frequency-selective channels. We consider the case where the receiver analog front-end suffers from IQ-imbalance and the local oscillator suffers from carrier-frequency offset (CFO). While the IQ-imbalance results in a mirroring effect, the CFO induces inter-carrier interference (ICI). In addition to ICI, we consider the channel delay spread is larger than the cyclic prefix (CP). This means that inter-block interference (IBI) is present. The frequency-domain equalizer is obtained by transferring a time-domain equalizer (TEQ) to the frequency-domain resulting in a per-tone equalizer (PTEQ). Due to the presence of IQ-imbalance the conventional TEQ (where only one TEQ is applied to the received sequence) is not sufficient to cope with the mirroring effect. A sufficient TEQ consists of two time-domain filters; one applied to the received sequence and another applied to a conjugated version of the received sequence. For the case of IQ-imbalance and CFO, the TEQs are designed according the basis expansion model (BEM) which showed to be able to cope with the ICI problem. Finally, in addition to the frequency-domain PTEQ design procedure, a training-based RLS type initialization scheme for direct per-tone equalization is also proposed     

Per-tone echo cancellation for DMT-based systems

A new echo cancellation structure for discrete multitone systems is presented, where each used tone has its own per-tone echo canceller in addition to a per-tone equalizer, which provides an alternative to currently employed time domain and time/frequency domain approaches. The per-tone approach enables us to optimize the signal-to-noise ratio for each tone separately by solving a minimum mean-square error problem for each tone, with implicit so-called joint shortening. Complexity during data transmission is compared for time domain, time/frequency domain, and per-tone echo cancellation. Structures with reduced complexity are derived for an interpolated and a decimated rate setup. Finally, simulation results for an asymmetric digital subscriber line setting demonstrate improved performance over time domain (or time/frequency domain) echo cancellation.     

Equalization for OFDM over doubly selective channels

In this paper, we propose a time-domain as well as a frequency-domain per-tone equalization for orthogonal frequency-division multiplexing (OFDM) over doubly selective channels. We consider the most general case, where the channel delay spread is larger than the cyclic prefix (CP), which results in interblock interference (IBI). IBI in conjunction with the Doppler effect destroys the orthogonality between subcarriers and, hence, results in severe intercarrier interference (ICI). In this paper, we propose a time-varying finite-impulse-response (TV-FIR) time-domain equalizer (TEQ) to restore the orthogonality between subcarriers, and hence to eliminate ICI/IBI. Due to the fact that the TEQ optimizes the performance over all subcarriers in a joint fashion, it has a poor performance. An optimal frequency-domain per-tone equalizer (PTEQ) is then obtained by transferring the TEQ operation to the frequency domain. Through computer simulations, we demonstrate the performance of the proposed equalization techniques.     

Per-tone equalization for MIMO OFDM systems

This paper focuses on multiple-input multiple-output (MIMO) orthogonal frequency division multiplexing (OFDM) systems with channel order larger than the cyclic prefix (CP) length. Writing the demodulating fast Fourier transform (FFT) as a sliding FFT followed by a downsampling operation, we show in this paper that by swapping the filtering operations of the MIMO channel and the sliding FFT, the data model for the temporally smoothened received signal of each individual tone of the MIMO OFDM system is very similar to the data model for the temporally smoothened received signal of a MIMO single-carrier (SC) system. As a result, to recover the data symbol vectors, the conventional equalization approach for MIMO SC systems can be applied to each individual tone of the MIMO OFDM system. This so-called per-tone equalization (PTEQ) approach for MIMO OFDM systems is an attractive alternative to the recently developed time-domain equalization (TEQ) approach for MIMO OFDM systems. In the second part of this paper, we focus on direct per-tone equalizer design and adapt an existing semi-blind equalizer design method for space-time block coding (STBC) SC systems to the corresponding semi-blind per-tone equalizer design method for STBC OFDM systems.     

Adaptive bit rate maximizing time-domain equalizer design for DMT-based systems

The classical discrete multitone receiver as used in, e.g., digital subscriber line (DSL) modems, combines a channel shortening time-domain equalizer (TEQ) with one-tap frequency-domain equalizers (FEQs). In a previous paper, the authors proposed a nonlinear bit rate maximizing (BM) TEQ design criterion and they have shown that the resulting BM-TEQ and the closely related BM per-group equalizers (PGEQs) approach the performance of the so-called per-tone equalizer (PTEQ). The PTEQ is an attractive alternative that provides a separate complex-valued equalizer for each active tone. In this paper, the authors show that the BM-TEQ and BM-PGEQ, despite their nonlinear cost criterion, can be designed adaptively, based on a recursive Levenberg-Marquardt algorithm. This adaptive BM-TEQ/BM-PGEQ makes use of the same second-order statistics as the earlier presented recursive least-squares (RLS)-based adaptive PTEQ. A complete range of adaptive BM equalizers then opens up: the RLS-based adaptive PTEQ design is computationally efficient but involves a large number of equalizer taps; the adaptive BM-TEQ has a minimal number of equalizer taps at the expense of a larger design complexity; the adaptive BM-PGEQ has a similar design complexity as the BM-TEQ and an intermediate number of equalizer taps between the BM-TEQ and the PTEQ. These adaptive equalizers allow us to track variations of transmission channel and noise, which are typical of a DSL environment.     

浅海水声通信中的间接自适应均衡算法

为了消除具有时变和多径特性的浅海水声信道带来的码间干扰,该文提出了一种间接自适应均衡算法。在发送端,设计了一种适合在浅海水声信道中传输的帧结构。在接收端,利用帧头进行快速信道估计和均衡器抽头系数的初始化,通过门限的设定,有选择地激活特定位置的抽头。并采用可变步长的LMS算法进一步减小误差,更好地均衡帧体数据。理论分析和仿真结果表明,该算法具有快速收敛性和较低的硬件复杂度,对经过高阶调制的发送信号具有很好的均衡效果,可以有效地应用于浅海高速数字通信系统。     

A new system identification method for fast echo cancelerinitialization

Proposes a new recursive version of an earlier technique for fast initialization of data-driven echo cancelers (DDECs). The speed of convergence and the covariance of the estimate of the proposed technique are comparable to the recursive least squares (RLS) algorithm, however, the computational complexity is no greater than the least mean square (LMS) algorithm. Analysis of computational complexity and the estimation error is also provided. Simulation results based on both floating-point and fixed-point arithmetic illustrate a remarkable improvement in terms of speed of convergence and steady-state error over the computationally comparable LMS algorithm     

Complex scaled tangent rotations (CSTAR) for fast space-timeadaptive equalization of wireless TDMA

A new update algorithm for space-time equalization of wireless time-division multiple access signals is presented. The method is based on a modified QR factorization that reduces the computational complexity of the traditional QR-decomposition based recursive least squares method and maintains numerical stability. Square roots operations are avoided due to the use of an approximately orthogonal transformation, defined complex scaled tangent rotation     

MIMO OFDM系统中的Turbo子载波均衡

在MIMO OFDM系统中,为了对抗同天线干扰及由于保护间隔不足而引起的码间干扰和载波间干扰,该文给出了一种基于MMSE的Turbo子载波均衡器。在该算法中,软输入软输出(SISO)的子载波均衡器与软输入软输出(SISO)解码器通过迭代进行软信息交换。仿真结果表明,与非迭代的子载波均衡器相比,该文给出的Turbo子载波均衡器能够有效利用时间和空间分集,使系统性能得到了改善。     

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。     

Reduced complexity short-block data detection techniques for fadingtime-dispersive channels

The objective is reliable digital communications over fading channels with severe time-dispersion. Three short-block data detection techniques based on linear, nonlinear decision-directed, and maximum likelihood estimation principles are investigated. Short alternating blocks of data and training symbols are used. The training blocks are useful for both channel tracking and data block detection. In contrast to the recursive symbol-to-symbol equalization approaches usually employed, each data block is detected as a unit. Previous restrictions imposed on short-block detection schemes have been removed, resulting in reduced complexity algorithms and higher throughput efficiency. Performance is presented for BPSK and QPSK type signaling using a Rayleigh fading channel model with severe time-dispersion     

Local convergence of the Sato blind equalizer and generalizationsunder practical constraints

An early use of recursive identification in blind adaptive channel equalization is an algorithm developed by Y. Sato (1975). An important generalization of the Sato algorithm with extensive analysis appears in the work of A. Benveniste et al. (1980). These generalized algorithms have been shown to possess a desirable global convergence property under two idealized conditions. The convergence properties of this class of blind algorithms under practical constraints common to a variety of channel equalization applications that violate these idealized conditions are studied. Results show that, in practice, when the equalizer is finite-dimensional and/or the input is discrete (as in digital communications) the equalizer parameters may converge to parameter settings that fail to achieve the objective of approximating the channel inverse. It is also shown that a center spike initialization is insufficient to guarantee avoiding such ill-convergence. Simulations verify the analytical results     

一种空时分组码在频率选择性信道下的半盲递推解码

该文针对频率选择性信道下的Alamouti空时分组码,提出了一种在接收端未知信道信息的情况下的半盲的递推解码方法。该方法基于子空间方法,利用少量的导频符号和空时分组码的结构,递推地同时完成信道均衡和解码,直接解得信息符号。与已有的按块批处理的盲解码方法相比,递推方法因为在递推过程中利用已经判得的过去的信息符号,能够方便地进行性能与运算量的折衷。仿真结果验证了方法的有效性。     

基于Kalman滤波的水声混合双向迭代信道均衡算法

水声信道均衡中基于信道估计的均衡方法理论上具有更优的均衡性能,但较高的计算复杂度限制了算法的实际应用。针对这一问题,该文首先基于Kalman滤波和Turbo均衡提出一种迭代Kalman均衡器,实现了基于软符号的迭代信道估计与迭代Kalman均衡,且复杂度较常规方法降低约1个数量级。其次,针对单一均衡算法和单一方向Turbo均衡器存在的误差传递现象,设计了基于迭代Kalman均衡器与改进成比例归一化LMS (IPNLMS)自适应均衡器相结合的混合双向Turbo均衡器,提高了自适应均衡器的收敛速度和均衡性能,并通过双向均衡结构带来的增益改善了符号估计误差传递的现象。理论分析与仿真实验验证了该文算法的有效性。     

Linear and decision-feedback per tone equalization for DMT-based transmission over IIR channels

The per-tone equalizer (PTEQ) has been presented as an attractive alternative for the classical time-domain equalizer (TEQ) in discrete multitone (DMT) based systems, such as ADSL systems. The PTEQ is based on a linear minimum mean-square-error (L-MMSE) equalizer design for each separate tone. In this paper, we reconsider DMT modulation and equalization in the ADSL context under the realistic assumption of an infinite impulse response (IIR) model for the wireline channel. First, optimum linear zero-forcing (L-ZF) block equalizers for arbitrary IIR model orders and cyclic prefix (CP) lengths are developed. It is shown that these L-ZF block equalizers can be decoupled per tone, hence they lead to an L-ZF PTEQ. Then, based on the L-ZF PTEQ, low-complexity L-MMSE PTEQ extensions are developed: the linear PTEQ extension exploits frequency-domain transmit redundancy from pilot and unused tones; alternatively, a closely related decision-feedback PTEQ extension can be applied. The PTEQ extensions then add flexibility to a DMT-based system design: the CP overhead can be reduced by exploiting frequency-domain transmit redundancy instead, so that a similar bitrate as with the original PTEQ is achieved at a lower memory and computational cost or, alternatively, a higher bitrate is achieved without a considerable cost increase. Both PTEQ extensions are also shown to improve the receiver's robustness to narrow-band interference.     

基于多尺度灰度变换的图像增强研究

为了提高图像增强效果,有效地保持原始图像细节,提出了一种基于多尺度灰度变换的图像增强方法。利用梯度域递归双边滤波对原始图像进行多尺度分解。基于小波变换,将分解层的子带分别作灰度变换。根据变换后的各个子带重构得到分解层的增强结果,并在其基础上实现图像的整体增强。对比直方图均衡化、灰度变换,提出的方法增强效果更好,并且保图像细节。利用客观性能指标对增强结果进行评价。实验结果表明,提出的方法有效,并具有结构简单,计算复杂度低的特点。     

Application of Sequential Estimation Algorithm Based on Branch Metric to Frequency-Selective Channel Equalization

In this work, a sequential estimation algorithm based on branch metric is used as channel equalizer to combat intersymbol interference in frequency-selective wireless communication channels. The bit error rate (BER) and computational complexity of the algorithm are compared with those of the maximum likelihood sequence estimation (MLSE), the recursive least squares (RLS) algorithm, the Fano sequential algorithm, the stack sequential algorithm, list-type MAP equalizer, soft-output sequential algorithm (SOSA) and maximum-likelihood soft-decision sequential decoding algorithm (MLSDA). The BER results have shown that whilst the sequential estimation algorithm has a close performance to the MLSE using the Viterbi algorithm, its performance is better than the other algorithms. Beside, the sequential estimation algorithm is the best in terms of computational complexity among the algorithms mentioned above, so it performs the channel equalization faster. Especially in M-ary modulated systems, the equalization speed of the algorithm increases exponentially when compared to those of the other algorithms.     

一种新的半盲联合信道数据估计算法

提出一种新的半盲联合信道估计与数据检测器,通过将递归-EM信道估计算法嵌入Turbo均衡器中,根据接收信号进行逐符号信道估计,不但能够适应时变信道,而且有效提高了联合信道数据估计收敛速度。仿真表明本算法对初始信道响应估计精度要求很低,从而大大降低了对训练序列长度的要求,使用很短的训练序列就能够达到较好性能,有效提高了频谱利用率。     

改进的杂波协方差矩阵结构估计方法

在球不变随机向量建模的非高斯杂波背景下,针对现有协方差矩阵估结构计方法对杂波不具备完全自适应性以及计算复杂度高等问题,本文利用经验信息,提出了基于经验的自适应估计方法(E-AE),并将其作为初始化矩阵,再次利用经验信息进行迭代估计,得到了基于经验的自适应迭代估计方法(E-ARE)。E-AE和E-ARE只需要进行实数运算,减小了计算复杂度。从理论上证明了所提方法对应的ANMF对杂波纹理分量和协方差矩阵结构都具有CFAR特性,并通过仿真实验对其有效性进行了验证。最后,利用匹配滤波器的输出信杂比损失及相应的ANMF检测性能,对所提方法和已有方法进行效果评估,结果表明,本文方法具有收敛速度快、所需的辅助单元数少、信杂比损失小和对应的ANMF检测性能优等特点。