太赫兹通信中的一种高速并行均衡算法及其FPGA实现

基于驰豫超前变换中的超前展开、求和近似和延时近似技术,提出了流水线并行自适应CMA盲均衡算法。利用基于迭代短卷积的并行FIR滤波算法分析了提出的并行自适应盲均衡算法的滤波部分的高效实现结构;再利用基于组合短卷积的并行自适应系数更新算法分析了提出的并行均衡算法的系数更新部分的高效实现结构,从而得到了基于短卷积的流水线并行自适应盲均衡的完整实现框图,并分析了各模块的流水线延时需满足的关系。最后对该并行自适应盲均衡算法进行了FPGA量化实现,并通过MATLAB仿真及实际FPGA实现结果的对比,验证了本并行均衡算法的正确性和有效性。     

Efficient processing of acoustic signals for high-rate information transmission over sparse underwater channels

For underwater acoustic channels where multipath spread is measured in tens of symbol intervals at high transmission rates, multichannel equalization required for bandwidth-efficient communications may become prohibitively complex for real-time implementation. To reduce computational complexity of signal processing and improve performance of data detection, receiver structures that are matched to the physical channel characteristics are investigated. A decision-feedback equalizer is designed which relies on an adaptive channel estimator to compute its parameters. The channel estimate is reduced in size by selecting only the significant components, whose delay span is often much shorter than the multipath spread of the channel. Optimal coefficient selection (sparsing) is performed by truncation in magnitude. This estimate is used to cancel the post-cursor ISI prior to linear equalization. Spatial diversity gain is achieved by a reduced-complexity pre-combining method which eliminates the need for a separate channel estimator/equalizer for each array element. The advantages of this approach are reduction in the number of receiver parameters, optimal implementation of sparse feedback, and efficient parallel implementation of adaptive algorithms for the pre-combiner, the fractionally-spaced channel estimators and the short feedforward equalizer filters. Receiver algorithm is applied to real data transmitted at 10 kbps over 3 km in shallow water, showing excellent results.     

Steepest-descent method-based adaptive frequency domain equalizer in coherent optical orthogonal frequency division multiplexing with quadrature phase shift keying modulation

An adaptive frequency domain equalizer (A-FDE) in quadrature phase shift keying-coherent optical-orthogonal frequency division multiplexing (QPSK-CO-OFDM) system is proposed in this paper. A-FDE uses steepest descent method to search the optimal equalization parameters. And through applying the adaptive control factor (ACF) to monitor the state of A-FDE, the interference from phase noise (PN) and chromatic dispersion (CD) can be mitigated. The numerical simulation shows that the performance of A-FDE has a remarkable improvement on combating PN and CD comparing with the conventional decision-directed equalizer (DDE). And the A-FDE can achieve the forward-error-correction (FEC) threshold in noisier scenario. Furthermore, because the calculation of ACF only bases on valid data instead of pilot symbols (PSs), A-FDE has low PS overhead merit.     

相干水声通信系统中均衡器长度设计方法

基于自适应均衡技术的相干通信被证明是有效的水声通信方式之一。在均衡器设计过程中,确定均衡长度的方法主要是经验确定与搜索算法,前者依赖于特定的环境,后者一般计算量较大。本文提出一种应用于相干水声通信系统的根据信道冲激响应确定均衡长度的设计方法。仿真及试验结果证明了该方法可以有效地提高通信的可靠性。     

单载波频域均衡中的水声信道频域响应与噪声估计

本文对单载波频域均衡(SC-FDE)中的信道和噪声估计进行研究,理论分析了形成误码率平台的可能原因,提出了基于Chu序列的联合信道时频域响应和噪声功率估计算法.利用水声信道响应的稀疏特性和门限确定信道能量集中区域,进行信道估计的去噪处理和噪声功率计算.借助于水声射线模型对所提出的算法进行了仿真,分析了噪声估计对SC-FDE性能的重要影响.仿真结果表明,所提出的信道和噪声估计算法可以有效减缓或消除SC-FDE中的误码率平台.     

水声通信加权反馈双向Turb均衡器

为了提升自适应双向Turbo均衡器的收敛速度及降低误比特率,提出了采用加权反馈的双向Turbo均衡算法。首先在单个均衡器反馈输入中采用后验均值与先验均值混合的反馈方案,有效提升一轮迭代中均衡器输出的准确性;其次通过后验均值与先验均值的加权合并作为另一均衡器反馈的非因果项输入,在提升反馈输入准确性的同时提升了数据的利用率;最后在权值迭代中采用优化的比例归一化最小均方算法,提升训练阶段均衡器收敛速度。千岛湖试验中,在同样3.75 kbps通信速率的情况下,该方法误比特率仅为传统双向Turbo均衡器的1/3。仿真和试验数据表明,均衡器要达到同样的误比特率,本方法所需迭代轮数更少,在时变信道中系统稳定性更好,误比特率更低,提升了水声通信效率。      

Application of functional link artificial neural network for mitigating nonlinear effects in coherent optical OFDM

Artificial Neural Networks are developed as an important technique for equalization and have been widely used to mitigate the nonlinear effects in coherent optical systems. For the compensation of nonlinearities in coherent optical orthogonal frequency division multiplexing technique, the most popular artificial neural network model is a multilayer perceptron (MLP), as it is able to perform complex mapping between input and output spaces with significant success. However due to the complexity of multilayer perceptron nonlinear equalizer (MLP-NLE) model training of neural network is difficult. To overcome computational complexity issues of MLP-NLE, a single neuron based functional link artificial neural network nonlinear equalizer (FLANN-NLE) has been developed in this paper. Better performance of an equalizer is attributed to the usage of aPSO-BP algorithm for training the FLANN-NLE. The proposed FLANN-NLE surpasses the existing works both in terms of Q-Factor and computational complexity. For a fiber length of 1000 km and at launch power of ?6 dBm, the improvement in Q-Factor is approximately equal to 3.3 and 1 dB in contrast to the previously reported values of approximately 3 and 0.7 dB at bit rate of 40 and 80 Gbps respectively.     

水声通信中基于信道辨识的盲turbo均衡方法

水声信道多途效应明显,造成接收信号存在严重的码间干扰(ISI,Intersymbol interference)。基于最小均方误差(MMSE,Minimum mean square error)准则的turbo均衡器级联了均衡和信道译码,能够有效去除ISI,并获得优良的性能。由于水声信道的时变性,传统MMSE-turbo均衡需要周期性的训练序列,以实现连续可靠的通信。训练序列虽然提高了通信的可靠性,但降低了信息的有效传输速率。因此,为提高通信效率,本文提出了一种盲turbo均衡方法,该方法通过引入新的盲信道辨识器来同时获得信道估计响应和已去除部分ISI的初步均衡输出信号,并为turbo均衡提供初始的响应参数和比特软信息。与水声通信中应用较多的盲判决反馈均衡器(DFE,Decision feedback equalizer)相比,海上实验结果证明本文提出的盲turbo均衡方法抗信道多途衰落的能力较强,并且与传统MMSE-turbo均衡相比无需训练序列,因此提高了信息的有效传输速率。     

Adaptive and efficient nonlinear channel equalization for underwater acoustic communication

We investigate underwater acoustic (UWA) channel equalization and introduce hierarchical and adaptive nonlinear (piecewise linear) channel equalization algorithms that are highly efficient and provide significantly improved bit error rate (BER) performance. Due to the high complexity of conventional nonlinear equalizers and poor performance of linear ones, to equalize highly difficult underwater acoustic channels, we employ piecewise linear equalizers. However, in order to achieve the performance of the best piecewise linear model, we use a tree structure to hierarchically partition the space of the received signal. Furthermore, the equalization algorithm should be completely adaptive, since due to the highly non-stationary nature of the underwater medium, the optimal mean squared error (MSE) equalizer as well as the best piecewise linear equalizer changes in time. To this end, we introduce an adaptive piecewise linear equalization algorithm that not only adapts the linear equalizer at each region but also learns the complete hierarchical structure with a computational complexity only polynomial in the number of nodes of the tree. Furthermore, our algorithm is constructed to directly minimize the final squared error without introducing any ad-hoc parameters. We demonstrate the performance of our algorithms through highly realistic experiments performed on practical field data as well as accurately simulated underwater acoustic channels.     

高均衡量微带宽带功率均衡器

 对用于行波管驱动级的高均衡量微带宽带功率均衡器进行了研究,提出了一种切实可行的电路拓扑。该电路利用并联谐振枝节形成带阻特性,用集总参数电阻实现了对衰减量的调节,形成类似陷波器的效应,用紧凑的结构实现了器件的小型化。针对一种X到Ku频段均衡量达到20 dB的增益曲线,采用上述拓扑结构,进行了仿真研究和实物制作,最终实现满足要求的宽带微波功率均衡器。该均衡器具有宽带宽、高均衡量和小体积的特点。     

水声通信系统中双向turbo均衡算法

提出了一种水声通信系统中直接自适应双向turbo均衡算法。摒弃了信道估计步骤,采用基于直接自适应的turbo均衡器,并利用内嵌数字锁相环的判决反馈均衡器结构跟踪时变信道,采用最速优化算法自适应调整迭代步长,使得收敛速度和算法性能得到很好折中。此外,利用最小均方误差准则,得到最优权重因子,对正向与反向turbo均衡结果加权求和,消除误差传播效应。仿真和湖上实验验证了方法的正确性,双向均衡的性能优于单向均衡。湖上实验结果表明,基于直接自适应算法相比于基于信道估计的算法,对时变信道不敏感,能获得更低的误比特率。      

Efficient equalization of multi-exciter distributed mode loudspeakers

An efficient digital equalization method is applied successfully to the problem of spectral equalization of multi-exciter distributed mode loudspeakers (DML). It is based on a chain of second-order sections of infinite impulse response parametric filters with very low computational cost. The method compensates for the measured multi-exciter DML response in order to achieve a desired frequency response. The sound radiation of these flat loudspeakers is a complex superposition of excited modes that vary strongly with frequency. Therefore, the characteristic multi-exciter DML spectrum is very irregular and is equalized with the method presented here for a natural, uncolored response. In multichannel systems, such as wave field synthesis (WFS), the use of efficient filters to equalize a large amount of drivers is an advantageous approach. The equalization process has been applied to two multi-exciter DML prototypes, comprising three and five exciters per panel. Both panel and exciter equalization have been addressed, which consequences on the filtered responses are discussed. Finally, some subjective assessments are carried out to optimize the order of the filter while maintaining the perceived quality of the equalization.     

Acoustic model for robustness analysis of optimal multipoint room equalization

In this paper, an acoustic model for the robustness analysis of optimal multipoint room equalization is proposed. The optimal multipoint equalization aims to have the optimal performance in a least-squares sense for all measured points. The model can be used for theoretical robustness estimation depending on the critical design parameters such as the number of measurement points, the distance between measurements, or the frequency before applying real equalization system. The analysis results show that it is important to set the appropriate number of measurement points and the distances between measurement points to ensure the enlarged equalization region at a specific frequency.     

An adaptive channel equalizer using self-adaptation bacterial foraging optimization

In this paper we propose a self-adaptation bacterial foraging optimization (SA-BFO) approach for an adaptive channel equalizer in which the weights of the equalizer are optimized to minimize the mean square error (MSE) and bit error rate (BER). The adaptive channel equalizer at the receiver removes or reduces the effects of inter symbol interference (ISI) and noise. Tests demonstrate that the proposed adaptive channel equalizer provides better convergence speed and minimal MSE and BER compared to a BFO and a normalized least mean square (NLMS) based equalizer.     

A study of intersymbol interference and transmission medium instability for an optical fibre system

A simple model is used to evaluate the power penalty accruing from nearest neighbour Intersymbol Interference (ISI) in a digital optical communications system using an integrate- and -dump decision process. The results compare closely with those obtained by Personick in a frequency domain analysis. An analysis of the effects of drift in the transmission bandwidth in the cable and of imperfect equalization indicates a simple design criterion for a fixed equalizer link.     

混沌通信系统中非线性信道的自适应组合神经网络均衡

针对混沌通信系统的非线性信道干扰问题,基于混沌信号重构理论和函数型连接神经网络理论,提出了一种横向滤波器与函数型连接神经网络组合(combination of transversal filter and functional link neural network,CFFLNN)的自适应非线性信道均衡器,并给出基于低复杂度归一化最小均方(NLMS)的自适应算法,并对该均衡器的稳定性以及收敛条件进行了分析.该非线性自适应均衡器充分利用了横向滤波器的快速收敛,以及函数型连接神经网络通过增大输入空间提高非线性逼近能力的特点,进一步提高均衡器的收敛速度和降低稳态误差.仿真研究表明：所提出的非线性自适应均衡器能够有效地消除线性和非线性信道干扰,均衡器输出信号能反映出混沌信号的特性,具有良好的抗干扰性能;且该均衡器的结构简单,收敛稳定性较好,易于工程实现. 关键词： 非线性信道 自适应均衡器 混沌吸引子 神经网络     

Experimental implementation of a low-frequency global sound equalization method based on free field propagation

An experimental implementation of a global sound equalization method in a rectangular room using active control is described in this paper. The main purpose of the work has been to provide experimental evidence that sound can be equalized in a continuous three-dimensional region, the listening zone, which occupies a considerable part of the complete volume of the room. The equalization method, based on the simulation of a progressive plane wave, was implemented in a room with inner dimensions of 2.70 m × 2.74 m × 2.40 m. With this method, the sound was reproduced by a matrix of 4 × 5 loudspeakers in one of the walls. After traveling through the room, the sound wave was absorbed on the opposite wall, which had a similar arrangement of loudspeakers, by means of active control. A set of 40 digital FIR filters was used to modify the original input signal before it was fed to the loudspeakers, one filter for each transducer. The optimal arrangement of the loudspeakers and the maximum frequency that can be equalized is analyzed theoretically in this paper. The presented experimental results show that sound equalization was possible from 10 Hz to approximately 425 Hz in the listening zone. A flat frequency response with deviations within ±5 decibels from the desired value was achieved. A higher demanding performance with deviations within ±1.5 decibels from a flat frequency response was attained in the interval between 20 Hz and 280 Hz. At the same time, the impulse response was quite well approximated to a delayed delta function in the listening zone. Examples of the spatial distribution of the sound field are also shown.     

Strategies for bass enhancement in Multiactuator Panels for Wave Field Synthesis

In this paper, an extension of the useful bandwidth in the low frequency band is applied to compensate the inherent poor bass response of Multiactuator Panels. These are a special type of flat panel loudspeakers that are commonly used to reproduce spatial audio under the Wave Field Synthesis system. The proposed algorithm combines two strategies: first, a dynamic electrical equalization, applied to additional exciters, which are carefully positioned to excite the lower frequency modes of the panel; second, a psycho-acoustical approach, taking profit of the behavior of human hearing based on the missing fundamental principle. For comparison purposes, the shelving and peak equalizations are also applied to the MAP prototypes developed in this paper. Objective and subjective results show that the combined approach results in an effective extension in the low frequency end of MAP with very low levels of distortion, outperforming conventional equalization methods, with better low frequency behavior, and hence better audio quality perception.     

偏振复用光纤通信系统色散均衡器及算法的研究

设计了一种适用于偏振复用相干解调光纤通信系统的色散均衡器,用于补偿信道传输的色散损伤。该均衡器采用半码元间隔的蝶形有限脉冲响应滤波器结构,与此结构配合的自适应算法分别采用最小均方算法和递归最小二乘算法。通过仿真实验,分析了两种算法对残留色度色散和偏振模色散的补偿容限。仿真结果表明,递归最小二乘算法的补偿效果优于最小均方算法,它可以同时补偿1760ps/nm的残留色度色散和104.9ps偏振模色散引起的差分群时延,比同等条件的最小均方算法提升性能2.23dB。     

Baseband equalization in fibre optic digital transmission

The problem of baseband equalization in optical p.c.m. fibre systems is considered. Zero-forcing, minimum-mean-square error, and Kaiman equalizers are briefly reviewed and their relative merits discussed. The performance of the overall communication system is the average probability of error, which can be computed by means of the proposed method with high accuracy. A final section presents some diagrams that may be useful in designing the equalizer and in evaluating the degradation affecting the error probability with respect to ideal conditions. A presentation ofadaptive equalization results concludes the paper.