A New Equalization Method for Lithium-Ion Battery Packs Based on CUK

Aiming at the traditional CUK equalizer can only perform energy equalization between adjacent batteries,if the two single batteries that need to be equalized are far away from each other,there will be the problem of longer energy transmission path and lower equalization efficiency,this paper optimizes the CUK equalizer and optimizes its peripheral selection circuit,which can support the equalization of single batteries at any two positions.The control strategy adopts the open-circuit voltage(OVC)of the battery and the state of charge(SOC)of the battery as the equalization variables,and selects the corresponding equalization variables according to the energy conditions of the two batteries that need to be equalized,and generates the adaptive equalization current with an adaptive PID controller in order to improve the equalization efficiency.Simulation modeling is performed in Matlab/Simulink 2021b,and the experimental results show that the optimized CUK equalizer in this paper improves the equalization time by 25.58%compared with the traditional CUK equalizer.In addition,compared with the mean value difference(MVD)method,the adaptive PID method reduces the equalization time by about 30%in the static and charging and discharging experimental environments,which verifies the superiority of this equalization scheme.

Design of a 6.25 Gbps backplane SerDes with adaptive decision feedback equalization

A 6.25 Gbps SerDes core used in the high speed backplane communication receiver has been designedbased on the OIF-CEI-02.0 standard. To counteract the serious Inter-Symbol-Interference (ISI),the core employed a half-rate four-tap decision feedback equalizer (DFE). The equalizer used the Sign-signleast mean-squared (SS-LMS) algorithm to realize the coefficient adaptation. An automatic gain control(AGC) amplifier with the sign least mean-squared (S-LMS) algorithm has been used to compensatethe transmission media loss. To recover the clock signal from the input data serial and provide for theDFE and AGC, a bang-bang clock recovery (BB-CR) is adopted. A third order phase loop lock (PLL)model was proposed to predict characteristics of the BB-CR. The core has been verified by behavioralmodeling in MATLAB. The results indicate that the core can meet the specifications of the backplane receiver.The DFE recovered data over a 34' FR-4 backplane has a peak-to-peak jitter of 17 ps, a horizontaleye opening of 0.87 UI, and a vertical eye opening of 500 mVpp.

Two Blind Adaptive Equalizers Connected in Series for Equalization Performance Improvement

A variable step-size parameter is usually used to accelerate the convergence speed of a blind adaptive equalizer with N1 + N2 -1 coefficients where N1 and N2 are odd values. In this paper we show that improved equalization performance is achieved when using two blind adaptive equalizers connected in series where the first and second blind adaptive equalizer have N1 and N2 coefficients respectively compared with the case where a single blind adaptive equalizer is applied with N1 + N2 -1 coefficients. It should be pointed out that the same algorithm (cost function) is used for updating the filter taps for the different equalizers and that a fixed step-size parameter is used. Simulation results show that for the low signal to noise ratio (SNR) environment and for the case where the convergence speed is slow due to the channel characteristics, the new method has a faster convergence speed with a factor of approximately two while leaving the system with approximately the same or lower residual intersymbol interference (ISI).

An improved least mean square/fourth direct adaptive equalizer for under-water acoustic communications in the Arctic

An improved least mean square/fourth direct adaptive equalizer(LMS/F-DAE)is proposed in this paper for underwater acoustic communication in the Arctic.It is able to process complex-valued baseband signals and has better equalization performance than LMS.Considering the sparsity feature of equalizer tap coefficients,an adaptive norm(AN)is incorporated into the cost function which is utilized as a sparse regularization.The norm constraint changes adaptively according to the amplitude of each coefficient.For small-scale coefficients,the sparse constraint exists to accelerate the convergence speed.For large-scale coefficients,it disappears to ensure smaller equalization error.The performance of the proposed AN-LMS/F-DAE is verified by the experimental data from the 9th Chinese National Arctic Research Expedition.The results show that compared with the standard LMS/F-DAE,AN-LMS/F-DAE can promote the sparse level of the equalizer and achieve better performance.

An Approximated Expression for the Residual ISI Obtained by Blind Adaptive Equalizer and Biased Input Signals

Recently, two expressions (for the noiseless and noisy case) were proposed for the residual inter-symbol interference (ISI) obtained by blind adaptive equalizers, where the error of the equalized output signal may be expressed as a polynomial function of order 3. However, those expressions are not applicable for biased input signals. In this paper, a closed-form approximated expression is proposed for the residual ISI applicable for the noisy and biased input case. This new proposed expression is valid for blind adaptive equalizers, where the error of the equalized output signal may be expressed as a polynomial function of order 3. The new proposed expression depends on the equalizer’s tap length, input signal statistics, channel power, SNR, step-size parameter and on the input signal’s bias. Simulation results indicate a high correlation between the simulated results and those obtained from our new proposed expression.

Soft Direct-Adaptation Based Bidirectional Turbo Equalization for MIMO Underwater Acoustic Communications

This paper proposes a soft direct-adaptation based bidirectional turbo equalizer for multiple-input multiple-output underwater acoustic communication systems. Soft, rather than hard, direct-adaptation based equalizer combined with the fast self-optimized least mean square algorithm is employed to achieve a faster convergence rate, and the second-order phase-locked loop is embedded into the equalizer to track the time-varying channel. Meanwhile, by utilizing a weighted linear combining scheme, the conventional soft direct-adaptation based equalizer is combined with the time-reversed soft direct-adaptation based equalizer to exploit bidirectional diversity and mitigate error propagation. Both the simulation and experimental results demonstrate that the soft direct-adaptation based bidirectional turbo equalizer outperforms the single-direction soft direct-adaptation based turbo equalizer, and achieves a faster convergence rate than the hard direct-adaptation based bidirectional turbo equalizer.

A NEW ADAPTIVE EQUALIZER STRUCTURE FOR NONLINEAR CHANNELS

Based on the analysis of nonlinear channel models,a new connectionist model ofadaptive equalizer is constructed.Comparing with the connectionist model using the Volterraseries to extend the input vector space,the number of weights with the new structure is reducedsignificantly.It is shown by simulations that the weight values of the new scheme converge to theoptimal values closely for non-minimum phase channels as well minimum phase channels,if thechannel noise is small enough.Testing results of the BER(Bit Error Rate)tell us that the newadaptive equalizer for nonlinear channels is superior to the conventional linear equalizers in theequalization performances.

Adaptive blind equalizer based on least square support vector machine

An adaptive blind support vector machine equalizer（ABSVME） is presented in this paper.The method is based upon least square support vector machine（LSSVM）,and stems from signal feature reconstruction idea.By oversampling the output of a LSSVM equalizer and exploiting a reasonable decorrelation cost function design,the method achieves fine online channel tracing with Kumar express algorithm and static iterative learning algorithm incorporated.The method is verified through simulation and compared with other nonlinear equalizers.The results show that it provides excellent performance in nonlinear equalization and time-varying channel tracing.Although a constant module equalization algorithm requires that the signal has characteristic of constant module,this method has no such requirement.

Convergence Curve for Non-Blind Adaptive Equalizers

In this paper a closed-form approximated expression is proposed for the Intersymbol Interference (ISI) as a function of time valid during the entire stages of the non-blind adaptive deconvolution process and is suitable for the noisy, real and two independent quadrature carrier input case. The obtained expression is applicable for type of channels where the resulting ISI as a function of time can be described with an exponential model having a single time constant. Based on this new expression for the ISI as a function of time, the convergence time (or number of iteration number required for convergence) of the non-blind adaptive equalizer can be calculated. Up to now, the equalizer’s performance (convergence time and ISI as a function of time) could be obtained only via simulation when the channel coefficients were known. The new proposed expression for the ISI as a function of time is based on the knowledge of the initial ISI and channel power (which is measurable) and eliminates the need to carry out any more the above mentioned simulation. Simulation results indicate a high correlation between the simulated and calculated ISI (based on our proposed expression for the ISI as a function of time) during the whole deconvolution process for the high as well as for the low signal to noise ratio (SNR) condition.

TRANSFORM DOMAIN CONJUGATE GRADIENT ALGORITHM FOR ADAPTIVE FILTERING

This paper proposed a new normalized transform domain conjugate gradient algorithm (NT-CGA), which applies the data independent normalized orthogonal transform technique to approximately whiten the input signal and utilises the modified conjugate gradient method to perform sample-by-sample updating of the filter weights more efficiently. Simulation results illustrated that the proposed algorithm has the ability to provide a fast convergence speed and lower steady-error compared to that of traditional least mean square algorithm (LMSA), normalized transform domain least mean square algorithm (NT- LMSA), Quasi-Newton least mean square algorithm (Q-LMSA) and time domain conjugate gradient algorithm (TD-CGA) when the input signal is heavily coloured.

Enhancement in Channel Equalization Using Particle Swarm Optimization Techniques

This work proposes an improved inertia weight update method and position update method in Particle Swarm Optimization (PSO) to enhance the convergence and mean square error of channel equalizer. The search abilities of PSO are managed by the key parameter Inertia Weight (IW). A higher value leads to global search whereas a smaller value shifts the search to local which makes convergence faster. Different approaches are reported in literature to improve PSO by modifying inertia weight. This work investigates the performance of the existing PSO variants related to time varying inertia weight methods and proposes new strategies to improve the convergence and mean square error of channel equalizer. Also the position update method in PSO is modified to achieve better convergence in channel equalization. The simulation presents the enhanced performance of the proposed techniques in transversal and decision feedback models. The simulation results also analyze the superiority in linear and nonlinear channel conditions.

一种宽带非线性信道校正方法

针对高速数传解调器接收宽带高阶调制信号时因信道非线性导致的性能恶化问题,提出一种基于两层自适应均衡网络的非线性信道校正方法。该方法先利用递归最小二乘法获得Volterra非线性前置校正系数,再利用自适应基带均衡器消除信道的剩余影响,以提升解调器的捕获跟踪能力,并在高速数传解调器工程应用中实现2.4 Gb/s 16QAM信号的稳定接收。

基于改进虚拟同步发电机的构网型并联储能逆变器控制研究

针对构网型变换器预同步过程时电力系统频率电压稳定性低导致变换器并网失败的问题,提出一种基于改进线性自抗扰控制(LADRC)的构网型变换器预同步控制策略。首先,采用无锁相环(PLL)控制策略,通过相位偏差的反馈控制对网侧电压的相位和幅值进行快速同步追踪,可避免因PLL精度较低和响应速度慢而引起的电力系统稳定性降低的问题。在此基础上,在有功频率支路模块的角频率输出端引入LADRC,这可以有效解决预同步控制过程中变换器输出电压频率存在过冲的问题,从而确保构网型变换器预同步控制顺利进行最终实现成功并网。最后,在MATLAB/Simulink仿真平台中搭建基于改进LADRC的构网型变换器预同步控制模型并进行仿真验证,结果表明所提控制策略可以有效抑制系统频率震荡,并能够加速系统预同步进程,确保构网型变换器安全运行最终实现成功并网,仿真结果验证了所提方法的有效性。

基于CLAHE-PCA的矿井低照度图像增强研究

地下矿山巷道环境往往面临光线不足,难以通过获取其暗通道图像判断岩体剥落等异常情况。针对矿井巷道暗通道图像对比度低的问题,提出了一种基于CLAHE-PCA的图像增强算法。首先使用CLAHE算法将获取的矿井巷道原始暗通道图像做对比度增强处理,然后使用自适应Gamma算法对亮度低的图像予以增加对比度矫正;将矫正后获得的灰度图转为RGB图像,通过PCA对其进行平滑处理,以便更多地还原暗通道图像的细节。以峰值信噪比、结构相似性、平均梯度和信息熵等作为评价指标,对试验结果进行验证。结果表明:该方法能够有效处理低对比度的矿井巷道图像,处理后的图像结构相似性达到93%,鲁棒性强,同时能够更多地还原图像的细节。

基于自适应模糊PID算法的锂电池组双层均衡控制

为提高电池重组时的均衡效率,在传统Buck-Boost均衡拓扑电路的基础上,设计了一种锂电池组双层均衡拓扑电路。组内采用Buck-Boost电路均衡,组间利用双向反激变压器进行均衡。均衡控制策略采用自适应模糊PID算法,以电池荷电状态(state of charge, SOC)为均衡变量,利用模糊控制算法对PID参数进行调节,缩短了均衡时间,提高了均衡效率。在Matlab/Simulink中搭建了锂电池组双层均衡拓扑电路和自适应模糊PID控制算法模型。实验结果表明:在不同工作状态下,所提出的电池组均衡拓扑及其控制策略将均衡时间效率平均提高了58.36%,验证了该方案的有效性。

基于自适应色彩均衡及改进IBLA的水下图像增强

针对水下环境采集图像和视频存在对比度低、色彩不均衡等颜色失真问题,提出了一种自适应色彩均衡与水下场景深度估计融合的水下图像增强算法。该算法的核心思想由3个步骤组成:首先针对图像色偏严重问题,对色偏严重的通道进行颜色自适应均衡化预处理,以突出纹理特征;其次,根据图像整体的纹理分布特性进行特征提取,并对图像通道进行背景光估计、场景深度估计以及传输系数映射估计;最后,采用场景亮度恢复模型进行特征融合,并基于直方图均衡算法得到水下增强图像。为证明该算法的有效性与先进性,选取真实水下降质图像为处理对象,利用质量评价指标(UIQM)对增强图像进行总体比较。结果表明,该算法有效增强了水下降质图像的清晰度与色彩度,并且清晰地恢复了图像细节。

面向56 Gb/s高速SerDes接收机DSP设计

高速接口芯片是高性能互连网络通信中的一款重要IP,针对56 Gb/s四脉冲幅度调制信号在高性能互连网络背板通信中,由于传输距离长信道衰减严重导致误码率高的问题,提出一种面向56 Gb/s高速Serdes接收机DSP设计。该DSP采用64路并行结构,通过16抽头前向反馈均衡器,以及1抽头预判决反馈均衡器对接收端数字化后的信号进行处理;采用基于K-均值聚类算法生成动态变化的判决电平并结合最小均方误差算法,能够处理15~35 dB不同信道衰减下的均衡问题。为了验证算法的性能,还搭建了一个基于模拟前端芯片和现场可编程门阵列的实验验证平台。实验结果表明,在信道衰减为15~35 dB@14 GHz,速率为56 Gb/s的条件下,误码率均小于5e-10。

一种面向112 Gb/s PAM4接收机的自适应均衡设计方案

提出了一种适用于超短距离(Very Short Reach,VSR)信道、面向112 Gb/s PAM4(Pulse Amplitude Modulation 4)接收机的自适应均衡设计方案。在该方案中,接收机前端利用3个连续时间线性均衡器(Continuous Time Linear Equalizer,CTLE)对信号分别在高频、中频和低频进行补偿,可变增益放大器(Variable Gain Amplifier,VGA)和饱和放大器(Saturation Amplifier,SatAmp)则用于对信号幅值的缩放。除了3个数据采样器外,引入4个辅助采样器用于进一步改善阈值自适应算法性能。同时,采用符号最小均方算法,利用接收端数据采样器和辅助采样器之间的偏移推动辅助参考电压收敛到信号星座电平,从而确保PAM4接收信号的眼图在垂直方向上3个眼睛具有相等的间隔和恒定的信噪比(Signal-to-Noise Ratio,SNR)。仿真结果表明,所提出的112 Gb/s PAM4接收机能够在损耗为15 dB的信道上实现小于10~(-12)的误码率,并且具有良好的眼图性能,其最差眼高为75 mV,眼宽为0.34 UI(Unit Interval),与传统方案相比具有显著的性能提升。

非正交多址光通信系统信道均衡的虚拟时反方法

室内非正交多址(non-orthogonal multiple access, NOMA)可见光通信系统(visible light communications, VLC)可实现高速率多用户通信,但多径效应会造成通信可靠性及用户公平性明显下降,为此,提出一种虚拟时间反转技术(virtual time reversal mirror, VTRM)对NOMA-VLC进行信道均衡,消除多径效应对通信过程的影响。分析了多用户场景下室内NOMA-VLC系统模型及其通信光链路的特性,依据光链路增益的稀疏特性采用稀疏度自适应匹配追踪(sparsity adaptive matching pursuit,SAMP)算法,估计NOMA-VLC系统的信道冲激响应(channel impulse response, CIR)。在此基础上,采用VTRM方法对NOMA-VLC进行信道均衡,通过VTRM的时空聚焦特性减少信道衰落影响,并重构接收信号,抑制多径效应。理论分析和实验结果表明:经论文算法均衡后的NOMAVLC系统,在满足前向纠错(forward error correction,FEC)误码率阈值的同时,用户1性能提高了4.4 dB,用户2性能提高了5.7 dB,两用户平均性能提高了5.05 dB,用户间性能差异由1.6 dB降为0.5 dB,为NOMA-VLC信道均衡提供了有效的解决方案。