用于盲接收的turbo均衡与迭代信道估计

Turbo均衡是一种将信道均衡和差错控制译码联合迭代处理的均衡机制,与传统的均衡器相比,能在更低的信噪比条件下克服严重信道失真导致的符号间干扰(ISI)。该文提出一种用于盲接收的turbo均衡和迭代信道估计方案,不依赖于训练序列或发送符号的先验知识,采用初始盲均衡处理来启动迭代信道估计和turbo均衡。在该方案中,初始盲均衡算法的选择是十分关键的,它需要在恶劣的信道条件为后续迭代处理的启动提供足够的先验信息。该文根据turbo均衡的特点,选择了超指数算法作为初始盲均衡。仿真结果表明,该文提出的用于盲接收的turbo均衡方案是有效的。     

Autocorrelation properties of channel encoded sequences-applicability to blind equalization

Many blind channel equalization/identification algorithms are derived assuming the transmitted information sequence to be white. In practical communication systems, redundancy is added to the source sequence in order to detect and correct symbol errors in the receiver. It is not obvious how channel encoding will affect the assumption of whiteness. The autocorrelation function of some commonly used channel codes is analyzed in order to study the validity of assumptions used in blind equalization. The codes are presented in terms of a Markov model for which the autocorrelation is analytically expressed. The various encoded sequences are used in a prediction error based blind equalizer, and the performance is empirically compared with the case of unencoded data. A blind equalization example using a practical GSM speech encoder combined with a convolutional channel encoder is also given.     

Blind detection of equalization errors in communication systems

In adaptive channel equalization, transmitted symbol estimates at the equalizer output may be in error because of excessive channel noise, convergence of the equalizer to a “closed-eye” local minimum, or error propagation if the equalizer has a decision feedback structure. This paper is concerned with the detection of equalization errors (i.e., errors in transmitted symbol estimates) in a blindfolded manner whereby no direct access to the channel input is required. The detection problem is cast into a binary hypothesis testing framework. Assuming a linear communication channel that is time-invariant during the test interval, a relationship between the presence of equalization errors and time variations in the underlying linear model taking the transmitted symbol estimates to the equalizer input is established. Based on this relationship, a uniformly most powerful test is constructed to detect the presence of equalization errors in finite-length observations. Finite sample size and asymptotic detection performance of the test is studied. A method for estimating the equalization delay without direct access to the channel input is developed. The effectiveness of the test is illustrated by way of computer simulations     

Differential modulation techniques for a 34 MBit/s radio channel using orthogonal frequency division multiplexing

The orthogonal frequency division multiplexing (OFDM) technique has been proposed for terrestrial digital transmission systems due to its high spectral efficiency, its robustness in different multipath propagation environments and the ability of avoiding intersymbol interference (ISI). Our studies consider a radio channel bandwidth of 8 MHz and a data rate of 34 Mbit/s.In the case of the OFDM transmission system a coherent 64-QAM requires a channel estimation process and a channel equalization in frequency-selective interference situations [4]. The equalization process can be realized by a multiplier bank at the FFT output in the receiver, a so-called frequency-domain equalizer. Alternatively, a multilevel differential modulation technique, the so-called differential amplitude and phase shift keying (64-DAPSK) considering the phase and simultaneously the amplitude for differential modulation, is proposed and presented in this paper. Differential modulation/demodulation techniques do not require any explicit knowledge about the radio channel properties in the differential channel equalization. It is therefore not necessary to implement a frequency-domain equalizer in an OFDM/64-DAPSK receiver, which reduces the computation complexity. The performance of both modulation techniques has been analysed in the uncoded and coded case referring to Gaussian and frequency-selective Rayleigh fading channels. Simulation results are presented in this paper.The OFDM signal has a non-constant envelope with large instantaneous power spikes possible primarily resulting in an overdriving of the high power amplifier (HPA) at the transmitter. This leads to nonlinear distortion causing intermodulation noise and spectral spreading. Both effects can be limited by introducing an appropriate input backoff (IBO). In this paper the performance of OFDM signals in the presence of nonlinearities is analysed quantitatively.     

基于叠加训练序列和低复杂度频域Turbo均衡的时变水声信道估计和均衡

针对时变水声信道估计和均衡问题,该文提出基于叠加训练序列(ST)和低复杂度频域Turbo均衡(LTE)的时变水声信道估计和均衡(ST-LTE)算法。基于叠加训练序列方案,将训练序列和符号线性叠加,使得训练序列和符号信道信息一致;基于最小二乘算法,进行信道估计。基于频域训练序列干扰消除技术,在频域消除训练序列对符号的干扰;基于频域线性最小均方误差(LMMSE)均衡算法,通过先验、后验、外均值和方差的计算,实现低复杂度信道均衡(符号估计);基于Turbo均衡算法,软重构叠加训练序列和更新信道估计,进行均衡器和译码器的信息交换,利用编码冗余信息,大幅度提升信道均衡性能。进行仿真、水池静态通信试验(通信频率12 kHz,带宽6 kHz,采样频率96 kHz,符号传输速率4.8 ksym/s,训练序列和符号的功率比为0.25:1)和胶州湾运动通信试验(通信频率12 kHz,带宽6 kHz,采样频率96 kHz,符号传输速率3 ksym/s,训练序列和符号的功率比为0.25:1),仿真和试验结果验证了所提算法的有效性。     

Self-Recovering Equalization and Carrier Tracking in Two-Dimensional Data Communication Systems

Conventional equalization and carrier recovery algorithms for minimizing mean-square error in digital communication systems generally require an initial training period during which a known data sequence is transmitted and properly synchronized at the receiver. This paper solves the general problem of adaptive channel equalization without resorting to a known training sequence or to conditions of limited distortion. The criterion for equalizer adaptation is the minimization of a new class of nonconvex cost functions which are shown to characterize intersymbol interference independently of carrier phase and of the data symbol constellation used in the transmission system. Equalizer convergence does not require carrier recovery, so that carrier phase tracking can be carried out at the equalizer output in a decision-directed mode. The convergence properties of the self-recovering algorithms are analyzed mathematically and confirmed by computer simulation.     

基于DDEA算法的短波信道Turbo均衡研究

针对数据引导均衡算法（DDEA）的缺陷,受到Turbo均衡中迭代思想的启发,提出一种基于数据引导均衡技术的短波迭代信道估计、均衡和译码算法,不仅均衡和译码之间互相传递软信息,信道估计器使用的也是译码器反馈的软信息,通过每次迭代时软信息质量的改善,提高系统的可靠性;通过多次迭代信道估计器输出的信道参数很接近当前帧的实际信...     

Use of blind equalization for teletext broadcast systems

A new application of blind equalization that appears to ameliorate the delays and improve the reliability of teletext systems is presented. The compatibility with the existing TV system and the cost-effectiveness of this application were considered for practical purposes. Equalizer adaptation does not require a knowledge of the transmitted data sequence or carrier phase recovery. Therefore, a reduction in the existing teletext circuitry has been achieved. In addition, the proposed quadrature phase shift keying (QPSK) modulation scheme doubles the teletext data transmission rate and uses the allocated TV bandwidth efficiently. The performance of the proposed blind equalizer has been evaluated in the presence of a multipath broadcast channel. Simulations have shown that substantial improvements are observed in the bit-error-rates for both two-phase PSK and QPSK modulation schemes     

MLSE and soft-output equalization for trellis-coded continuousphase modulation

This paper presents two equalizer structures for trellis-coded continuous phase modulation (TC-CPM) on multipath fading intersymbol interference (ISI) channels. An equivalent discrete-time (DT) model is developed by combining the tapped-delay-line (TDL) model of the frequency-selective channel and by oversampling at the receiver. The (noninterleaved) fractionally spaced maximum-likelihood sequence estimation (MLSE) equalizer performs continuous phase modulation (CPM) demodulation, trellis-coded modulation (TCM) decoding, and channel equalization by exploiting the finite state nature of the ISI-corrupted TC-CPM signal. Both simulation and analytical results show diversity-like improvement when performing joint MLSE decoding and equalization. For the interleaved soft-output equalizer, the soft symbol metric is delivered to the TCM decoder by using a forward and backward recursion algorithm. Three variants of the soft-output equalizer are examined. We conclude that the backward recursion is essential to partial response CPM schemes, and with moderate complexity, the soft-output equalizer can have a substantial advantage over a noninterleaved MLSE equalizer     

DVB-C接收机中自适应均衡器的新设计

着重讨论一种全新的DFE与盲均衡相结合的自适应均衡器的设计,对于符合DVB—C标准的QAM信号,在输入存在多径、频偏及加性噪声的情况下,采用此设计无须借助于训练序列或导频即可成功对信道进行自适应均衡以抵消ISI,并能自动切换捕获和跟踪模式,仿真效果良好。此外还能在一定范围内对输入的符号功率进行调整以降低AGC模块的设计要求。     

频率选择性衰落OFDM的半盲信道估计算法研究

从系统性能分析和设计的角度详细地研究了基于无线HIPERLAN2通信协议的OFDM(orthogonal frequency division multiplexing，正交频分多路复用技术)系统信道估计与均衡的一种半盲算法，提出了结合直接法和Cholesky分解法的切换盲算法。这种半盲算法综合了全盲算法得到的信息与已知导频符号，充分利用了原发信号的统计特性和OFDM帧结构中插入的导频符号，克服了全盲和导频训练序列存在的问题^[6,7]，且不需额外的带宽。仿真结果表明，在误比特率和收敛性方面，该算法比现有的主要三种全盲算法有更好的收敛和抗干扰特性。     

Blind equalization and channel estimation with partial responseinput signals

The problem of blind equalization and channel estimation for partial-response signals (PRS) is considered. Three approaches are investigated; two of them exploit the prior knowledge of the PR code, and the third approach does not. We propose a constrained optimization approach involving a quadratic cumulant matching criterion where the coding structure of the transmitted signal is assumed to be a priori known. The other two approaches exploit the Godard blind equalizer. Computer simulation results using 16-QAM signals, and duobinary and modified duobinary PR codes, show that the constrained optimization approach yields the best performance as measured via the probability of symbol detection error     

Semi-Blind Time-Domain Equalization for MIMO-OFDM Systems

In this paper, a semi-blind time-domain equalization technique is proposed for general multiple-input-multiple-output (MIMO) orthogonal frequency-division multiplexing (OFDM) systems. The received OFDM symbols are shifted by more than or equal to the cyclic prefix (CP) length, and a blind equalizer is designed to completely suppress both intercarrier interference (ICI) and intersymbol interference (ISI) using second-order statistics of the shifted received OFDM symbols. Only a one-tap equalizer is needed to detect the time-domain signals from the blind equalizer output, and one pilot OFDM symbol is utilized to estimate the required channel state information for the design of the one-tap equalizer. The technique is applicable irrespective of whether the CP length is longer than, equal to, or shorter than the channel length. Computer simulations show that the proposed technique outperforms the existing techniques, and it is robust against the number of shifts in excess of the CP length.     

Joint equalization and decoding for QAM block transmissions

A method of joint equalization and decoding for block transmission with coding across blocks is presented. It is well suited for bandwidth efficient QAM transmission using large transmit symbol alphabets, whereby the blocking structure can be arbitrary in principle. The method consists in multiple iterations of equalization and decoding, whereby information is exchanged back and forth between the decoder and the equalizer. The equalizer is based on Cholesky factorization of the equivalent discrete-time channel matrix (Eur. Trans. Telecommun 10 (1999) 351; Grundlagen der Kommunikationstechnik, Springer, Berlin, 2004). The convergence behavior of the iterative equalization and decoding method is analyzed by means of an upper bound on the extrinsic information transfer. It is shown that the method has improved bit error performance versus separate equalization and decoding without decoder feedback and, thus, yields superior power efficiency.     

高速突发传输中的均衡算法

通过对几种常用均衡算法的比较,确定高速突发传输中使用收敛速度较快、运算量较小的VLMS均衡算法。为降低高速突发信号中的训练序列开销,提高传输效率,提出了一种新的混合结构均衡方案。新方案充分利用数据辅助均衡与盲均衡的各自特点,将均衡过程分成两个阶段,数据辅助均衡阶段为盲均衡阶段提供一个足够精确的初始值,引导盲均衡实现快速收敛。仿真结果表明,该混合结构均衡方案能利用短训练序列实现收敛,而且能达到较好的性能。     

Blind equalization for short burst wireless communications

In this paper, we propose a dual mode blind equalizer based on the constant modulus algorithm (CMA). The blind equalizer is devised for short burst transmission formats used in many current wireless TDMA systems as well as future wireless packet data systems. Blind equalization is useful for such short burst formats, since the overhead associated with training can be significant when only a small number of bits are transmitted at a time. The proposed equalizer overcomes the common problems associated with classic blind algorithms, i.e., slow convergence and ill-convergence, which are detrimental to applying blind equalization to short burst formats. Thus, it can eliminate the overhead associated with training sequences. Also, the blind equalizer is extended to a two branch diversity combining blind equalizer. A new initialization for fractionally spaced CMA equalizers is introduced. This greatly improves the symbol timing recovery performance of fractionally spaced CMA equalizers with or without diversity, when applied to short bursts. Through simulations with quasi-static or time-varying frequency selective wireless channels, the performance of the proposed equalizer is compared to selection diversity and conventional equalizers with training sequences. The results indicate that its performance is far superior to that of selection diversity alone and comparable to the performance of equalizers with short training sequences. Thus, training overhead can be removed with no performance degradation for fast time-varying channels, and with slight performance degradation for static channels     

On channel estimation and sequence detection of interleaved codedsignals over frequency nonselective Rayleigh fading channels

Due to the receiver complexity introduced by interleaving, the implementation of maximum likelihood (IML) decoding of interleaved coded signals transmitted over frequency nonselective Rayleigh fading channels has been shown to be practically impossible. As an alternative, a two-stage receiver structure has been proposed, where the channel estimation and sequence decoding are done separately. The channel estimation issue in a two-stage receiver is examined in detail in this paper. It is shown that although an optimum (maximum a posteriori (MAP)) channel estimation is not possible in practice, it can be approached asymptotically by joint MAP estimation of the channel and the coded data sequence. The implementation of the joint MAP estimation is shown to be an ML sequence estimator followed by an minimum mean-square error (MMSE) channel estimator. Approximate fill sequence estimation using pilot symbol interpolation is also studied, and through computer simulations, its performance is compared to receivers using hit sequence estimation. The effect of decision delay (DD), prediction order, and pilot insertion rate (PIR) on the reduced complexity ML sequence estimation is investigated as well. Finally, a practical receiver is proposed that makes the best compromise among the error performance, receiver complexity, DD, and power (or bandwidth) expansion due to pilot insertion     

利用CAZAC序列的OFDM频率同步方法

频率偏移估计是OFDM (Orthogonal Frequency Division Multiplexing)的关键技术。该文提出了在一个OFDM符号内利用两个重复的CAZAC (Constant Amplitude Zero Auto Correlation)序列进行频率偏移估计的新方法,估计范围可达整个OFDM带宽。用于同步的训练序列还可用于信道估计和均衡,这将提高系统的数据传输效率。此算法同步精度达到了Schmidl提出的Cramer-Rao bound,有效性在AWGN和Rayleigh多径信道下得到了验证。     

一种不受信道阶数估计影响的SIMO直接盲均衡算法

针对单输入多输出（Single-Input-Multiple-Output, SIMO）模型提出一种完全不需要信道阶数估计的直接盲均衡算法。文章利用接收数据的截短协方差矩阵和信号子空间的关系设计一种零延迟均衡器,并通过信道矩阵和均衡器系数的合响应特性克服了算法相位偏转的问题,最后得到一种对信道阶数估计鲁棒并且没有相位偏转的盲均衡算法。该算法不同于一般子空间类算法,不需要直接对接收信号的协方差矩阵进行信号子空间和噪声子空间的分解,因此对信道阶数估计具有很强的鲁棒性。文章给出了算法的Batch实现过程,同时为更好适应一般时变信道环境和实现实时处理的要求,通过递归迭代得到算法的自适应实现过程。仿真实验表明该算法几乎不受信道阶数过估计或欠估计的影响,同时该算法具有良好的均方误差（Mean Square Error, MSE）和误符号率SER（Symbol Error Rate, SER）性能,并且具有很快的收敛速度。      

Adaptive maximum-likelihood sequence estimation by means ofcombined equalization and decoding in fading environments

This paper proposes an adaptive maximum-likelihood sequence estimation (MLSE) by means of combined equalization and decoding, i.e., adaptive combined MLSE, which employs separate channel estimation for respective states in the Viterbi algorithm. First, an approximate metric including channel estimation is derived analytically for this proposed adaptive combined MLSE. Secondly, procedures to accomplish blind equalization are investigated for the proposed MLSE. Finally, its excellent BER performance on fast time-varying fading channels is confirmed by computer simulation, when the proposed MLSE operates as a blind equalizer