Digital Cancellation Scheme and Hardware Implementation for High-Order Passive Intermodulation Interference Based on Hammerstein Model

Passive intermodulation(PIM)interference urgently needs to be solved in the satellite communication system,owing to degrading the whole performance.Mainstream research contributions to the cancellation method for PIM were focused on the analog domain,however,the PIM distortion cannot be eliminated completely with the approaches.Meanwhile,some researchers attempt to tackle the problem through digital signal processing,nevertheless,the proposed methods were not suitable for the practical satellite communication scenario.In this paper,we present a general scheme for the adaptive feedforward PIM cancellation.High-order PIM signals at baseband are estimated by modeling the PIM distortion with Hammerstein model in the digital domain.Based on the reconstructed PIM signal,we adopt the least mean square algorithm to adaptively mitigate the PIM interference for tracking the variation of PIM.The time and frequency synchronization of PIM are based on the correlation of the peak of received signals with the corresponding reconstructed PIM signal.Practical experimental results show that the scheme can effectively cancel the PIM interference,and achieve an interference suppression gain more than 20dB.

Adaptive digital self-interference cancellation based on fractional order LMS in LFMCW radar

Adaptive digital self-interference cancellation(ADSIC)is a significant method to suppress self-interference and improve the performance of the linear frequency modulated continuous wave(LFMCW)radar.Due to efficient implementation structure,the conventional method based on least mean square(LMS)is widely used,but its performance is not sufficient for LFMCW radar.To achieve a better self-interference cancellation(SIC)result and more optimal radar performance,we present an ADSIC method based on fractional order LMS(FOLMS),which utilizes the multi-path cancellation structure and adaptively updates the weight coefficients of the cancellation system.First,we derive the iterative expression of the weight coefficients by using the fractional order derivative and short-term memory principle.Then,to solve the problem that it is difficult to select the parameters of the proposed method due to the non-stationary characteristics of radar transmitted signals,we construct the performance evaluation model of LFMCW radar,and analyze the relationship between the mean square deviation and the parameters of FOLMS.Finally,the theoretical analysis and simulation results show that the proposed method has a better SIC performance than the conventional methods.

Full-duplex prototype based on joint passive and digital cancellation method

Recent research shows that it is possible to achieve the full-duplex system by cancelling strong self-interference signals, which can be divided into three classes respectively, i.e., passive cancellation, active cancellation and digital cancellation. This pa- per tries to achieve the full-duplex system without using active cancellation, thus a full-duplex system using a joint mechanism based on a novel passive cancellation method and a novel digital cancellation method is proposed. Therein, a good antenna place- ment guided by the theory of the antenna electromagnetic field for the passive cancellation is presented. For the proposed digital can- cellation method, unlike previous separate mechanisms, it is de- signed by using the recursive least square （RLS） algorithm jointly with passive cancellation. The self-interference channel state in- formation （CSI） is transferred as the input of digital cancellation to balance the performance and the complexity. Experimental results show that the proposed self-interference cancellation mechanism can achieve about 85 dB which is better than the previous re- search. Meanwhile, this design provides a better performance compared with half-duplex with both line-of-sight channel and non- line-of-sight channel.

Application of pole-zero cancellation circuit in nuclear signal filtering and shaping algorithm

In radiation measurement and digital nuclear spectrum systems,traditional nuclear signal processing circuits in nuclear electronics have been gradually replaced by digital algorithm modules with the application of highperformance programmable hardware logic devices(such as FPGA or DSP).Referring to the digital realization method of inverse RC integral circuit systems,the function of the pole-zero cancellation(PZC)circuit was analyzed,a new modified cascade equivalent model of PZC was established,and the time-domain digital PZC(DPZC)recursive algorithm was derived in detail in this study.Two parameters kIand k_(D)are included in the new algorithm,where kIshould match the exponential decay time constant of the input signal to realize the pole-zero compensation,while the decay time constant of the output signal can be changed with the adjustable parameter k_(D)(which is larger than the decay time constant of the input signal).Based on the new DPZC algorithm module,two trapezoidal(triangular)shaping filters were designed and implemented.The amplitude–frequency characteristics of the output signal of the proposed trapezoidal shaping algorithm and the convolution trapezoidal shaping algorithm were compared,with fixed peaking time.The results show that the trapezoidal shaping algorithm based on DPZC can better suppress high-frequency noise.Finally,based on the Na I(Tl)scintillator(u75 mm×75 mm)detector and^(137)Cs source,the effect of the k_(D)value on the energy resolution of the DPZC trapezoidal(triangular)shaping algorithm was studied.The experimental results show that,with an increase in k_(D),the energy resolution of the system improved and reached the maximum when k_(D)was greater than 10,and the optimal energy resolution of the system was 7.72%.

Minimally Invasive Maxillofacial Surgery Using Digital Work Surgery: A Case of Alveolar Ridge Reconstruction after Maxillary Cystectomy

Introduction: Iliac particulate cancellous bone and marrow (PCBM) is commonly used as a high-quality reconstruction material;however, PCBM cannot be extracted in sufficient amounts to meet demand. To determine the appropriate amount of iliac PCBM to be collected, we used digital technology to measure the volume required for jaw reconstruction before surgery. Clinical Case: The patient, a 23-year-old man, underwent surgery for a calcifying odontogenic cyst. A maxillary cyst occupied the left anterior-premolar region (tooth 21 - 25) and the deciduous canine remained;a permanent canine was included in the cyst. We planned to preserve the teeth except for the impacted canine, completely excise the maxillary cyst, and preserve the alveolar ridge morphology. Preoperative digital imaging was used to determine the amount of alveolar ridge reconstruction required and accordingly determine the amount of iliac cancellous bone to be harvested. We used a titanium mesh tray and grafts of iliac particulate cancellous bone and marrow to reconstruct the alveolar ridge. The amount of iliac cancellous bone that needed to be collected was clarified and the supply amount could be collected in just the right amount;thus, the cortical bone of the iliac inner plate could be preserved. The alveolar bone morphology was reconstructed to allow the placement of dental implants as per the preoperative digital surgery. Three years after the operation, no sign of recurrence has been observed. Conclusion: Minimally invasive surgery was performed by clarifying the amount of iliac cancellous bone graft that needs to be harvested, which improved the accuracy of surgery.

面向脉冲调制器的S波段高效线性MMIC功率放大器

针对脉冲调制全数字发射机中高速数字信号难以高效驱动射频功率放大器的问题,提出了一种高效线性MMIC功率放大器设计方法。采用连续F类功率放大器提高数字发射机系统的效率,采用三阶互调抵消改善临近饱和线性度。基于南京电子器件研究所GaAs pHEMT工艺,研制了一款S波段高效率线性功率放大器进行验证。测试结果表明,在3.4~4.0 GHz频段内,当输入功率为-10 dBm时,输出功率为26.2~26.7 dBm,功率附加效率为53.3%~60.7%,功率增益为36.2~36.7 dB。在3.7 GHz频点,输出功率为24.5 dBm,三阶互调失真小于-30 dBc;9 dB峰均比、40 MHz正交频分多路复用的64-QAM调制信号激励下,平均输出功率及对应的功率附加效率分别为20.5 dBm和28%,实现了-30 dBc的邻道功率比及5.5%的误差向量幅度。

基于改进鲸鱼算法的数字助听器回声消除方法

在数字助听器应用中,回声消除能力是检验助听器效果的标准之一。针对回声消除算法中自适应滤波器权重设置主观性较强以及实际语音信号波动性较大导致数字助听器回声消除精度不高的问题,提出了一种结合集合经验模态分解(EEMD)方法及鲸鱼算法(WOA)的最小均方误差算法(LMS)。该新方法在鲸鱼算法的基础上引入集合经验模态分解方法,采用集合经验模态分解方法将采集到的语音信号分解为若干个平稳的子序列,运用鲸鱼算法收敛速度较快的特点帮助自适应滤波器更快找到更适宜的权重,对自适应滤波器进行权重向量的优化,从而建立EEMD-WOA-LMS模型,通过回声消除效果的常用性能指标对所提方法进行验证。仿真实验表明,将改进方法与WOA-LMS算法相比,能获得较快的收敛速度和较低的失调量,并且均方误差有所降低,失调量降低了0.73 dB,回声损失值提升了2.87 dB,验证了EEMD-WOA-LMS算法的回声消除能力更优。

一种具有纹波消除技术的10 bit SAR ADC

逐次逼近寄存器模数转换器(SAR ADC)在逐次逼近的过程中,电容的切换会使参考电压上出现参考纹波噪声,该噪声会影响比较器的判定,进而输出错误的比较结果。针对该问题,基于CMOS 0.5μm工艺,设计了一种具有纹波消除技术的10 bit SAR ADC。通过增加纹波至比较器输入端的额外路径,将参考纹波满摆幅输入至比较器中;同时设计了消除数模转换器(DAC)模块,对参考纹波进行采样和输入,通过反转纹波噪声的极性,消除参考纹波对ADC输出的影响。该设计将信噪比(SNR)提高到56.75 dB,将有效位数(ENOB)提升到9.14 bit,将积分非线性(INL)从-1~5 LSB降低到-0.2~0.3 LSB,将微分非线性(DNL)从-3~4 LSB降低到-0.5~0.5 LSB。

一种基于FVF动态缓冲器的二阶噪声整形SAR ADC

本文提出了一种高精度的二阶无源噪声整形逐次逼近型模数转换器(NS SAR ADC)。首先,采用了一种通过动态浮动反向放大器(FIA)实现kT/C噪声消除的技术,随后结合差分结构的定制电容,实现更小的电容阵列面积的同时抑制了采样热噪声。最后,采用翻转电压跟随器(FVF)结构作为动态缓冲器提取残差电压,并通过电容堆叠的操作实现无源求和,对比较器噪声与量化噪声进行了二阶整形。该设计采用0.18μm工艺实现,仿真表明,在1.8 V电源电压、8倍过采样率和2 MS/s的采样频率下,信号噪声失真比(SNDR)为88.57 dB,无杂散动态范围(SFDR)为99.09 dB,功耗仅为302μW。

A novel analog/digital reconfigurable automatic gain control with a novel DC offset cancellation circuit

An analog/digital reconfigurable automatic gain control （AGC） circuit with a novel DC offset cancel- lation circuit for a direct-conversion receiver is presented. The AGC is analog/digital reconfigurable in order to be compatible with different baseband chips. What＇s more, a novel DC offset cancellation （DCOC） circuit with an HPCF （high pass cutoff frequency） less than 10 kHz is proposed. The AGC is fabricated by a 0.18μm CMOS process. Under analog control mode, the AGC achieves a 70 dB dynamic range with a 3 dB-bandwidth larger than 60 MHz. Under digital control mode, through a 5-bit digital control word, the AGC shows a 64 dB gain control range by 2 dB each step with a gain error of less than 0.3 dB. The DC offset cancellation circuits can suppress the output DC offset voltage to be less than 1.5 mV, while the offset voltage of 40 mV is introduced into the input. The overall power consumption is less than 3.5 mA, and the die area is 800 ×300μm2.

Digital and analog hybrid control in adaptive interference cancellation system

The digital proportion control is introduced to improve the performance of the analog adaptive interference cancellation system （ICS）. For the high frequency parts of the signals after multiplier are not required,the sampling frequency need not satisfy the sampling theorem for high frequency. Because the sampling,calculation and output expend time in digital control,the ideal condition,delay condition and delay-wait condition are taken into account. Through analyzing the system model with three conditions,we gain the stable conditions of the system,the optimization step factors that can make the system converge fastest and the formulas of the interference cancellation ratios （ICRs）. One step convergence can be accomplished under ideal condition,whereas the system can not converge in one step under delay condition and delay-wait condition. The calculation results show the convergence speed of delay-wait condition is slower than that of delay condition. The ICR is improved with the increase of the step factor which is in stable bound,but the convergence speed is decreased if the step factor exceeds the optimization step factor. In order to avoid that confine,the method of amending the steady state weight to improve the ICR is proposed. The analyses are in agreement with the computer simulations.

基于数字化设计仿真的射频干扰抵消SiP设计

随着无线电子系统朝着阵列化、微系统化、高频化等方向发展,系统调试难、失效、故障难以排除等问题逐渐凸显,基于“经验设计+后续调试”的传统设计方法已难以满足实际需求。基于机、电、热、磁等多学科数字化协同设计仿真能够对各设计阶段进行指导与验证,在提高设计效率的同时保证产品的性能指标,是微系统的主流设计方法。基于此,文中以射频干扰抵消SiP为例,详细介绍电、磁、热协同设计仿真的整个设计流程。实验表明,所提设计达到了预期目标,可为从事硬件电路设计者提供指导与参考。

基于部分解耦合的非线性数字干扰对消算法

针对数字对消器中基于并行Hammerstein(parallel Hammerstein,PH)模型的非线性最小均方(least mean square,LMS)算法基函数之间的强相关性而存在收敛速度变慢、收敛稳定性变差、稳态误差增大的问题,提出了一种采用部分解耦合策略的改进算法。通过引入带约束的正则方程,实现各个非线性阶次独立的迭代步长设定。并且对基于部分解耦合干扰对消算法进行解析分析,给出了平均和均方意义下的收敛条件。仿真结果表明,所提方法在几乎不损失稳态误差性能的前提下,提升了非线性LMS算法收敛速度,改善了收敛性能,并且针对滤波器非线性阶数和记忆长度变化具有良好的鲁棒性。在1 MHz带宽二进制相移键控(binary phase shift keying,BPSK)干扰和加性高斯白噪声情况下干噪比设置为20 dB,相较于传统非线性LMS算法,所提算法可以实现最大0.5倍迭代步长上界、最高非线性阶次9以及最大记忆长度12的设定。

核脉冲信号数字自动极零设计

针对数字极零相消参数调试过程中依赖人工干预的问题,在推导出负指数衰减脉冲信号时间常数计算模型基础上,提出了自动极零算法,搭建了数字自动极零软硬件测试装置。验证实验表明,提出的负指数衰减脉冲信号时间常数计算模型,可应用于核脉冲信号时间常数的精确计算,数字自动极零算法可直接应用于数字化多道核脉冲信号自动极零相消功能的实现。

无线传能磁通信带内干扰抑制射频架构与算法

针对磁耦合无线电能与数据协同传输系统中磁耦合数据通信面临的传能装置谐波干扰,建立了基于DD线圈的通信和干扰感测相互隔离的射频通道,并在数字域采用基于递归最小二乘(Recursive Least Square,RLS)算法的数字干扰对消,抑制通信链路的残余传能干扰。通过仿真验证了方案的可行性。建立了中心频率17.6 MHz、通信带宽2 MHz的通信与干扰感测通道,并基于GNU Radio和X310软件无线电平台完成了算法验证,成功恢复出受传能谐波干扰的正交相移键控(Phase-Shift Keying,QPSK)通信信号星座图,将带内干扰谐波抑制了12 dB以上,提升了磁耦合通信链路抗干扰能力。

数字阵列雷达辅助通道优化配置研究

为了研究辅助通道选择对二维数字阵列雷达旁瓣对消性能的影响,本文对二维数字阵列旁瓣对消原理进行了研究,对可能出现的主辅通道相关性差及空间模糊等问题进行了分析,总结了辅助通道的选取原则,并通过仿真验证了这些原则在实际场景下的可行性。

基于模数混合子带划分的信号重构建模分析与优化

为了应对宽带阻塞式干扰,通信干扰对消系统通常应用基于子带划分的干扰对消技术来提升宽带干扰抑制能力,同时为了兼顾通信性能,将子带信号重构恢复为通信信号,保障通信质量。该文采用模拟电路与数字处理相结合的方式,搭建了子带划分与信号重构框架,通过模拟滤波器实现第1级宽滤波,减小信号处理带宽,利用数字滤波器完成第2级窄滤波,进一步提高信号信噪比。针对通信信号在子带划分过程中,存在跨子带重构失真的问题,建立了子带划分与信号重构系统时频域模型,分析子带间幅相不一致对信号重构的影响。为了解决重构信号的幅相不一致问题,提出了相位校准方法与滤波器幅频优化方法,从幅度和相位两方面同时保证跨子带信号的近似无失真重构。仿真与实验结果表明,该文所设计的滤波器幅频响应具有良好的重构精度,并且解决了跨子带信号重构的相位失真问题,有效降低了重构通信信号的误码率。

主动降噪耳机数字滤波器设计研究

介绍了数字式主动降噪耳机的工作原理、总体实现方案以及其中的数字滤波器架构,重点研究了用于主动降噪系统的数字滤波器设计原理和数字滤波器单元的类型、性质及设计方法,分析了采用数字降噪方案的头戴式耳机的实测降噪性能结果。

Performance Evalution of 2B1Q Data Echo Canceller in Digital Subscriber Access Networks

Recently, a lot of digital subscriber loop systems (DSL) and high rate digital subscriber loop systems (HDSL) have been deployed in digital subscriber access networks. Data echo canceller is extensively used in the DSL and HDSL systems to realize full duplex transmission on twisted cable. In such systems, 2B1Q line code is adopted. Therefore, symbol rate is decreased and digital transmission systems get longer reach. In this paper, a performance evaluation method of the echo canceller is proposed based on an autocorrelation matrix of 2B1Q line code. Using this method, the formula of ratio of data signal to echo residual signal is obtained. According to the formula, the ratio of data signal to echo noise depends on FIR filter tap, convergence factor, adaptive algorithm and the correlation matrix of the line code. Computer simulation is carried out to verify theoretical analysis. The simulation results coincided with the theoretical formula in the process estimating the ratio of signal to noise.

数字阵列雷达在空域抗干扰方面的优势和局限性分析

分析了数字阵列雷达在空域抗有源干扰方面的优势,提出了一种适用于地面数字阵列雷达系统的综合抗干扰方法,该方法基于同时接收数字多波束的处理架构,能够在抗干扰的同时改善对低空飞行目标的测角精度。文中结合外场试验结果和实测数据进行了验证分析,还从现代雷达电子战的角度对数字阵列雷达的空域抗干扰的局限性进行了分析,以期望对新体制数字阵列雷达在抗干扰方面的优势和面临的挑战有一个更加全面的认识。