Frequency response of sampled-data systems
自动控制理论词汇表

Glossary for Feedback Control of Dynamic Systems自动控制理论词汇表Chapter 1thermostat n.恒温器predictive control 预测控制power generation plant 发电厂micron n.微米cell phone 移动电话jumbo jet 巨型喷气式客机block diagram 方框图actuator 执行机构process n.过程feedback n.反馈plant n.被控对象mph=mile per houropen-loop 开环closed-loop 闭环throttle n.油门gain n.增益orifice n.孔、小孔controlled variable 被控变量error n.误差incubator n.孵化器flue n.烟道chronicle n.编年史、年代记录conical a.圆锥体的mill wright 技工、造水车工匠inertia n.惯性、惯量oscillate about…在….周围振荡reference input 参考输入prescribed direction 预定的方向actual direction 实际方向flyball n.飞球governor n.控制器、调节器、总督、省长equilibrium n平衡点differencial equation 微分方程characteristicequation 特征方程third-order 三阶polynomial n.多项式state variable 状态变量distortion n.畸变complex variable 复变量methdology n.方法论proportional a. 比例的、成比例的integral a.积分的derivative a.微分的stochastic a.随机的servomechanism n.伺服机构calculus n.微积分ubiquitous a.到处存在的、普遍存在的radar-tracking 雷达跟踪SISO systems 单输入单输出系统Laplace transform 拉普拉斯变换pole n.极点zero n.零点transfer function传递函数trajectory optimization 轨迹优化root locus 根轨迹specifications n.指标、规格、规范discrete-data 离散数据sampled-data 采样数据performance n.性能Chapter 2desired reference variable 期望参考变量prototype n.原型system identification 系统辨识time response 时域响应step input 阶跃输入defer n.推迟、延期vector n.向量、矢量slug n.斯(勒格),质量单位(32.2磅)impart n.赋予、传授、告知heavy line 粗实线dashed line 虚线coordinate n.坐标numerator n.分子denominator n.分母suspension n.悬架、悬挂deflection n.偏移、偏转、偏差displacement n.位移shock absorber 减震器、缓冲器dashpot n.缓冲器bump n. vt.颠簸bounce n. vt.反弹、跳跃moment of inertia 转动惯量、惯性矩attitude n.姿态、姿势antenna n.天线perpendicular n.垂直线 a.垂直的asymmetry a.不对称的torque n.转矩resonant n.谐振、共振damper n.阻尼器prudent a.谨慎的,有远见的,精打细算的anti-alias 抗混频operational amplifer 运算放大器passive circuit 无源电路Kirchhoff’s current law 基尔霍夫电流定律algebraic sum 代数和summer n.加法器integrator n.积分器tesla n.特斯拉(磁通量单位)louderspeaker n.扬声器bobbin线轴,线筒stator n.定子rotor n.转子back emf 反电势maze n.曲径,迷宫specific heat 比热spatially ad.空间地hydraulic a.液压的、水力学的gimbal n.平衡环,万向接头nozzle n.喷嘴grooming n.修饰,美容piston n.活塞porous a.可渗透的,多孔的laminar a.多层的、层流的n.层流turbulent a.湍流的Chapter 3linear time-invariant systems 线性时不变(定常)系统signal flow graph 信号流图simulation n.仿真frequency-response 频率响应superposition n.叠加convolution n.卷积inpulse-response 脉冲响应unit step function 单位阶跃函数root-locus 根轨迹stability properties 稳定性特性principles linear algebra 线性代数原理state variable methods 状态变量法matrix n..矩阵nonlinear n.非线性mathematical mode 数学模型trivial a.琐碎的、不重要的linearize vt.线性化operating point 工作点state-space 状态空间partial differential equations 偏微分方程equilibrium n.平衡点complex frequency variables 复频率变量zero initial conditions 零初始条件steady-state 稳态ramp input 斜坡输入dc gain 直流增益inverse Laplace transform 逆拉氏变换partial fraction expansion 部分分式展开rantional a.有理的residue n.余式unilateral a.单边的convergence n.收敛final-value theorem 终值定理homogeneous differential equation 齐次微分方程ordinary differential equation 常微分方程overall transfer function 总的传递函数“loading” effect 负载效应cascade blocks 方框串联(级联)to reduce 化简eliminating 消去equivalent a.等效的simplification n.化简、简化integrodifferential a.积分-微分的time constant 时间常数imaginary axis 虚轴damping ratio n.阻尼比natural undamped frequency 自然无阻尼频率overdamped a.过阻尼critically damped n.临界阻尼rectangular coordinate 直角坐标oscillatory a.振荡的transient response 瞬态响应overshoot n.超调量delay time 延迟时间peak time 峰值时间rise time 上升时间settling time 调节时间steady state 稳态characteristic equation 特征方程RHP(Right Half-Plane) 右半平面elevator n.飞机升降舵,飞机升降仪,电梯nonminimum-phase 非最小相位diverge v.发散、分歧asymptotically stable 渐进稳定stability n.稳定性absolute stability 绝对稳定性relative stability 相对稳定性stability criterion 稳定性判据equilibrium state 平衡状态product n.乘积coefficient n..系数nagtive feedback 负反馈positive feedback 正反馈unity feedback system 单位负反馈系统reduction n.化简simultaneous a.联立的common factor 公因子expedient a.权宜的,有用的attenuate v.变弱,衰减,变细,变薄,稀释cofactor n.公因子Routh stability criterion 劳斯稳定性判据determinant n. 行列式tune v.调节retune v.再调节pseudorandom-noise 伪随机噪声signal-to-noise ratio 信-噪比Mason Gain Formula 梅森(增益)公式term n.术语signal flow graph 信号流图nodepathenvelope n.包络线dominant root 主导极点Chapter 4steady-state 稳态with respect to 关于….deviation n.偏离steady-state error 稳态误差load torque 负载转矩viscous friction 粘性摩擦repeater n.中继器drift v.漂移fidelity n.准确性,忠实,忠诚parabolic antenna 抛物线天线position error constant. 位置误差常数velocity error constant 速度误差常数robust property 鲁棒性shaft n.轴tachometer n.转速计inductance n.电感sampled v.采样quantized v.量化extrapolate v.预测,推测trapezoid n.梯形,不等边四边形vertices n.顶点order n.阶次,数量级proportional control 比例控制derivative control 微分控制sinusoidal a.正弦parameter n.参数Chapter 5root-locus method 根轨迹法monic a.首一的feedforward a.前向的denominator n.分母numerator n.分子quadratic n.二次项branch n.分支factored a.分解的asymptote vt.渐进n.渐进线division n.除法vantage point 有利地位,观点imaginary part 虚部breakaway point 分离点common denominator 公分母conjugate pairs 共扼对multiplicity n.多重,多数trial and error 凑试(法)spirule n.螺旋尺intersection n.交汇symmetrical a.对称的magnitude condition 幅值条件angle condition 相角条件phase condition 相角条件origin n.起始点terminus n.终点angle of departure 分离角、出发角angle of arrival汇合角、到达角cubic a.三阶的、立方的quartic a.四次的remainder n.留数、余数remainder theorem 留数定理taking the limit 取…极限synthetic division 综合除法dominant root 主导根compensator n.补偿器azimuth n.地位角、地平经度inertial guidance 惯性导航constant term 常数项symmetrical with respect to 关于…对称trial point 试验点terminate vt.终止于first differentiate 一阶微分real parts实部imaginary part 虚部lag compensator 滞后补偿器lead compensator 超前补偿器spill over 溢出,无法容纳autopilot n.自动导航trim v.n.使整齐,微调trim tab 平衡调整片margin n.裕量iteration n.重复、循环、迭代intact a.完好无缺的,原封不动的Chapter 6frequency response 频率响应rendered v.使成为,提供,报答,着色; 执行ratio of the magnitudes 幅值比bandwidth n.带宽resonant peak 谐振峰值low-pass filter低通滤波器sanity n.神智健全,头脑清楚,健全tangent n.正切、切线reciprocal a.互补的,相互的,互惠的phase difference 相角差transport lag 传输延迟irrational factor 非有理因子phase shift 相位移动moduli n.模(复数)poke vt.戳、刺、捅drudgery n.苦工、单调乏味的工作logarithmic coordinate 对数坐标semilog n.半对数decibel n.分贝decade n.十倍量程octave n.八倍频程、八度、八阶asymptotic behavior 渐进行为dotted n.虚线break frequency 转折频率corner frequency 转折频率slope n.斜率20dB/decade 20分贝/十倍频程superimpose vt.迭加polar plot 极坐标图pass function 旁路函数servomotor-amplifier 伺服电机-放大器angular velocity 角速度minimum phase 最小相位tilt angle 倾斜角lateral force 侧面力、横向力perceived velocity 可察觉的速度croseover frequency 穿越频率appendage n.附件、备件Nyquist criterion 奈奎斯特判据Semi-graphical 半图形Nyquist plot 奈奎斯特图Bode diagram 伯德图positive real part 正实部necessary and sufficient condition 充分必要条件left half of the s-plane s平面左半平面formidable a. 可怕的、令人生畏的determinant 行列式pole-zero cancellation 零极点相消rational functions 有理函数quotient n. 商、份额multi-loop control system 多环控制系统encircled vt. 环绕enclosed vt. 包围closed path 闭合路径counterclockwise a.逆时针的clockwise a.顺时针的encirclement n. 环绕enclosure n. 包围contour n.围线,轮廓线argument principle 幅角原理complex variable 复变量single-valued rational function 单值有理函数analytic a.解析的Nyquist path 奈奎斯特路径singularity n.奇异(点、值)semicircle n.半圆artifice n.技巧、技能gain margin 增益裕量phase margin 相角裕量vicinity n.邻近compromise n.折中,妥协trapezoidal a.梯形的iterate v.重复、循环、迭代bracket v.放在括号内,归入一类,包含octave n.八个一组的事物,八度enumerate v.数,点detrimental a.有害的,不利的threshold n.阈值Chapter 8sampling n.采样sample period 采样周期aliasing n.混频,别名inherent a.内在的z transform Z变换radar tracking system 雷达跟踪系统discrete period 离散周期discrete equivalent 离散等效digitization n.数字化recursive a.递归的,循环的difference equation 差分方程sample rate 采样速率sampler 采样器zero-order holder 零阶保持器inverse Z transform 反Z变换、逆Z变换long division 长除法unit circle 单位圆overlap n.重叠rephrase v.重新措辞,改述extrapolate v.预测,推测alleviate v.减轻,使 ... 缓和judicious a.明智的,贤明的,审慎的fictitious a.假想的,虚伪的impulse transfer function 脉冲传递函数piecewise-continuous 分段连续的pseudo-continuous-time 准连续时间Pade approximation Pade 近似Fourier analysis 傅立叶分析modulation n.调制Fourier transform 傅立叶变换spurious a.寄生的、伪的、假的ideal sampler 理想采样器impulse train 脉冲列、脉冲串transcendental a.超自然的、超常的rational function 有理函数closed form 封闭形式degree n.阶denominator n 分母numerator n 分子initial value 初始值identical a.相等的starred a.打星号的impulse response transfer function 脉冲响应传递函数uniformly spaced 均匀分布map into 影射、映射circles of radius 圆弧multiple-sheeted surfaceRiemann surface 黎曼曲面radial ray 射线by virtue of 借助、凭借、依靠….(的力量)logarithmic spiral 对数螺旋intersection n.相交power series 幂级数sampling instant 采样时刻natural logarithm 自然对数rationalizing 有理化cascading property 串联(级联)特性attenuation factor 衰减因子warp vt.使弯曲、使变形tune vt.调节、调整cross-hatched vt.用交叉线画出(图画上)阴影performance specification 性能指标trial-and-error approach 试凑法bilinear n.双线性Chapter 9equilibrium point 平衡点neighborhood n.邻域saturate n.饱和robotic n.机器人学heuristic a.启发式的,搜索式的sinusoidal a.正弦的sinusoid n.正弦harmonic a.谐波的describing-function 描述函数static nonlinearity 静态非线性dynamic nonlinearity. 动态非线性periodic response 周期响应phase-plane 相平面catastrophe n.灾难、浩劫shaky a.不稳定的,不可靠的scalar function 标量函数Liapunov function 李亚普诺夫函数linearization n.线性化inverse nonlinearity 可逆非线性perturbation n.摄动operating point 工作点eigenvalue n.特征值bearing n.轴承levitate v.浮动,使漂浮,使悬浮turbo n.汽轮机deviation n.偏差rigid link 刚性连接regime n.情形,体制dead-zone 死区viscous friction 粘性摩擦coulomb friction 库仑摩擦relay n.继电(特性)limit cycle 极限环deflect v.使偏,使歪windup n.终结,结束akin a.同类的,相似的odd function 奇函数backlash n.齿轮间隙magnetic hysteresis 磁滞coincident a.重合的,一致的on/off system 通断(控制)系统superposition n. 迭加sub-harmonic a.谐波的magnetic flux 磁通iron-cored coil 铁芯线圈stiction n. 静摩擦力autonomous a.自治的hypersphere n. 超球stability in the sense of Liapunov 李亚普诺夫意义下的稳定性asymptotically stable 渐进稳定monotonically stable 单调稳定origin n. 原点globally stable 全局稳定locally stable 局部稳定electronic oscillator 电子谐振器Van der Pol’s differential equation 范德波尔微分方程nonsinusoidal waveform 非正弦波形rated voltage 额定电压phase variable 相变量phase portrait 相图perpendicular a. 垂直的、正交的、成直角的Taylor series 泰勒级数increment n.增量Euler method 欧拉法singular point 奇异点。
频率响应FRA测试中的噪声分析

频率响应FRA测试中的噪声分析频率响应FRA是一种检测电力变压器绕组变形的测试方式,与其它现场检测方式相类似,频率响应FRA方式也容易受到现场噪声的影响。
噪声会模糊干扰一些重要的测试信息,这将会影响到对频率响应FRA结果的评估。
因此,了解噪声的来源、影响和抑制方式是非常必要的。
标签:变压器;FRA测试;噪声1.频率响应FRA方式频率响应FRA 方式用于在变压器发生故障之前,检测出变压器绕组的几何变形。
需要注意的是,本文讨论的FRA方式是扫频式的FRA,即SFRA方式(Sweep Frequency Response Analysis ),而非过去所用的脉冲式IFRA方式(Impulse Frequency Response Analysis)。
这是因为,相较于低压脉冲方式(IFRA),扫频方式(SFRA)在现场具有更好的重复性,因此,目前使用越来越广泛。
从图一可以看出,在变压器线圈的一端输入一个变频的正弦电压信号“U”,并从此点测量参考信号“U1”,与此同时,测量线圈的另一端的输出或响应信号“U2”。
这样,便可计算出传递函数H(f),表达式为(1)。
这意味着H(f)仅取决于频率响应FRA仪器的测量阻抗Rm和变压器阻抗Ztra。
图二是常见的频率响应FRA测试波形,对于大多数的测试而言,都是对频率响应的幅值图进行分析与评估。
不过,频率响应的相位图也具有一定的参考价值,图二的左下部分为相位图。
幅值的计算依据公式(2),相位的计算依据公式(3)。
2.FRA方式中的噪声介绍噪声定义为有害的干扰信号,它可能被添加在一个想获得的有用信号上。
噪声往往会模糊有用信号的信息内容,因此,噪声的检测与降低是很有必要的。
与任何其它的电气诊断方式一样,在现场,频率响应FRA的测试结果也会受到噪声的影响。
了解噪声的来源、影响与抑制方式是非常重要的,特别是比较不同厂家制造的频率响应FRA仪器时。
本文分析了频率响应FRA仪器的技术规格和噪声抑制能力的关系,并通过在电力变压器上进行的频率响应FRA测试实例来说明。
球-杆系统的非线性问题

Control, Harbin Engineering University, Harbin 150001 DOI: 10.1360/aas-007-0550
式 中 球 的 转 动 惯 量 Ib = 4.32 × 10−5 kg·m2, 球 的 质 量 m = 0.27 kg, 球 的 半 径 r = 0.02 m, 杆 的 转 动 惯 量 Iw = 0.1402 kg·m2, x 为球在杆上的位移, α 为杆的转角.
1 引言
在现代控制理论中, 球 – 杆系统是一个典型的实验装置, 国内不少高校也已配备这种实验装置. 近年来还有作者将球 – 杆系统作为一种典型的非线性系统来进行研究[1∼3]. 但是 实际调试发现球 – 杆系统并不是像文献中的数学模型所描述 的那么单纯. 实际的球 – 杆系统确实是一个非线性系统, 但 其中的非线性特性却尚未见有报道. 本文结合实测和调试, 对球 – 杆系统的非线性进行分析和说明. 只有了解了该系统 中的实际的非线性问题, 才能正确确定控制规律中的各个参 数, 使实验系统发挥正常的作用.
图 4 滞环特性 Fig. 4 Hysteresis
图 5 自振荡的描述函数法分析 Fig. 5 Describing function analysis
K d1 = [25.0418 26.9373 56.1205 6.1019]
ቤተ መጻሕፍቲ ባይዱ
(5)
注意到滞环特性的输入输出都是连续的, 而这个系统是 采样系统, 需要将此采样系统的回路在 α 处断开, 计算在连 续的正弦信号 θ 作用下的连续输出 α 之间的频率响应. 所以 这里采用 [6] 的方法来计算这个采样系统的频率响应 G(jω), 如图 5 所示. 这里 G(jω) 就是描述函数法分析中的系统线性 部分的特性.
C8051键盘测试程序

#include <c8051f000.h>#include <intrins.h>//----------------------------------------------------------------------------- #define uchar unsigned char#define uint unsigned int#define BAUDRATE#define SYSCLKvoid SYSCLK_Init (void);void delaynus(unsigned int q) ;void PORT_Init (void);void SPI0_Init (void);void LCD_Init(void);void SendSPIByte(unsigned char ch);void delaynms (unsigned int j);void writecom(unsigned char com);void writedata(unsigned char d);void writechar(unsigned char ua);void Write_COM(uchar ins);void lcden(datad);void LCD_set_xy( unsigned char x, unsigned char y );void LCD_write_string(unsigned char n);void lcd_key1(void);void lcd_key2(void);void lcd_key3(void);void lcd_key4(void);void UART0_Init (void);void presskey(void);//----------------------------------------------------------------------------- // Global CONSTANTS//----------------------------------------------------------------------------- sbit S3=P1^0;sbit S4=P1^1;sbit S5=P1^2;sbit S6=P1^3;sbit lcdcs=P3^0;unsigned char comd,kk,sdf,ppca;unsigned char virt_port,v,b,m;unsigned char lcd_data_count;unsigned char *lcdpoint;unsigned char qqq;unsigned char data8;unsigned int i;//----------------------------------------------------------------------------- //-----------------------------------------------------------------------------void main(void){WDTCN = 0xde; // disable watchdog timerWDTCN = 0xad;SYSCLK_Init ();PORT_Init ();UART0_Init() ; // initialize crossbar and GPIOSPI0_Init ();LCD_Init() ;delaynms (20);delaynus (100);LCD_set_xy(0X00,0);delaynus (200);presskey();delaynus (200);LCD_write_string(16);delaynus (200);while(1){if(S3==0){Write_COM(0X01);delaynms (200);LCD_set_xy(0X00,0);delaynus (200);lcd_key1();delaynus (200);LCD_write_string(6);delaynus (200);}else if(S4==0){Write_COM(0X01);delaynms (200);LCD_set_xy(0X00,0);delaynus (200);lcd_key2();delaynus (200);LCD_write_string(6);delaynus (200);}else if(S5==0){Write_COM(0X01);delaynms (200);LCD_set_xy(0X00,0);delaynus (200);lcd_key3();delaynus (200);LCD_write_string(6);delaynus (200);}else if(S6==0){Write_COM(0X01);delaynms (200);LCD_set_xy(0X00,0);delaynus (200);lcd_key4();delaynus (200);LCD_write_string(6);delaynus (200);}else{}}}//-----------------------------------------------------------------------------// Initialization Subroutines//-----------------------------------------------------------------------------//-----------------------------------------------------------------------------// PORT_Init//-----------------------------------------------------------------------------void PORT_Init (void){XBR0 = 0x27; // XBAR0: Initial Reset ValueXBR1 = 0x00; // XBAR1: Initial Reset ValueXBR2 = 0x5c; // XBAR2: Initial Reset ValuePRT0CF = 0x14; // Output configuration for P0PRT1CF = 0x10; // Output configuration for P3PRT3CF = 0x01; // Output configuration for P3}//-----------------------------------------------------------------------------// SYSCLK_Init//-----------------------------------------------------------------------------void SYSCLK_Init (void){OSCXCN = 0x67; // start external oscillator withfor (i=0; i < 256; i++) ; // XTLVLD blanking interval (>1ms)while (!(OSCXCN & 0x80)) ; // Wait for crystal osc. to settleOSCICN = 0x88; // select external oscillator as SYSCLK }//-----------------------------------------------------------------------------// SPI0_Init//-----------------------------------------------------------------------------void SPI0_Init (void){SPI0CFG = 0x07; // data sampled on 1st SCK rising edge SPI0CFG|=0xC0; //CKPOL =1;SPI0CN = 0x03; // Master mode; SPI enabled; flagsSPI0CKR = SYSCLK/2/2000000-1; // SPI clock <= 8MHz (limited by// EEPROM spec.)}//-----------------------------------------------------------------------------// UART0_Init//-----------------------------------------------------------------------------void UART0_Init (void){SCON = 0x50; // SCON: mode 1, 8-bit UART, enable RX TMOD = 0x20; // TMOD: timer 1, mode 2, 8-bit reload TH1 = -(SYSCLK/BAUDRATE/16); // set Timer1 reload value for baudrate TR1 = 1; // start Timer1CKCON |= 0x10; // Timer1 uses SYSCLK as time basePCON |= 0x80; // SMOD = 1TI = 1; // Indicate TX ready}//-----------------------------------------------------------------------------// LCD_Init//-----------------------------------------------------------------------------void LCD_Init(void) //向LCD送命令{// unsigned int xdata x;delaynms(100);datad=0x00;SendSPIByte(datad);delaynms(10);Write_COM(0x30);delaynms(10);Write_COM(0x30);delaynms(10);Write_COM(0x30);delaynms(10) ;Write_COM(0x28);delaynms(100);virt_port=0;SendSPIByte(virt_port);lcden(virt_port);Write_COM(0x01);delaynms(100);Write_COM(0x06);delaynms(10) ;Write_COM(0x0C);delaynms(500) ;}//----------------------------------------------------------------------------- // SendSPIByte//----------------------------------------------------------------------------- void SendSPIByte(unsigned char ch){ lcdcs=1;delaynus(100);SPIF = 0;SPI0DAT = ch;while (SPIF == 0);delaynus(100);lcdcs=0;delaynus(100);_nop_(); // 等待写结束}//----------------------------------------------------------------------------- // lcden//----------------------------------------------------------------------------- void lcden(datad){datad|=0x08;SendSPIByte(datad);datad&=0xf7;SendSPIByte(datad);}//----------------------------------------------------------------------------- // delaynms//----------------------------------------------------------------------------- void delaynms (unsigned int uu){unsigned int oo,ll;for (oo=0;oo<uu;oo++){for(ll=0;ll<1140;ll++);}}//----------------------------------------------------------------------------- // writechar//----------------------------------------------------------------------------- void writechar(unsigned char ua){uint j;uchar t,x;for(j=0;j<500;j++);datad|=0x02;SendSPIByte(datad);datad|=ua&0xf0;SendSPIByte(datad);datad|=0x08;SendSPIByte(datad);for(x=0;x<3;x++);datad&=0xf7;SendSPIByte(datad);for(x=0;x<3;x++);datad&=0x07;delaynus(100);SendSPIByte(virt_port);t|=ua&0x0f;datad|=t<<4;SendSPIByte(datad);for(x=0;x<3;x++);datad|=0x08;SendSPIByte(datad);for(x=0;x<3;x++);datad&=0xf7;SendSPIByte(datad);for(x=0;x<3;x++);datad=0x00;t=0x00;SendSPIByte(datad);}//----------------------------------------------------------------------------- // Write_COM//----------------------------------------------------------------------------- void Write_COM(uchar ins){uchar t;uint j;for(j=0;j<5000;j++); //用延时代替查询virt_port|=ins&0xf0;SendSPIByte(virt_port);//LCDE=1;virt_port|=0x08;SendSPIByte(virt_port);for(i=3;i>0;i--);virt_port&=~0x08;SendSPIByte(virt_port);virt_port&=0x07;SendSPIByte(virt_port);t=ins<<4;virt_port|=t&0xf0;SendSPIByte(virt_port);virt_port|=0x08;SendSPIByte(virt_port);for(i=3;i>0;i--);virt_port&=~0x08;SendSPIByte(virt_port);virt_port=0;SendSPIByte(virt_port);}//-----------------------------------------------------------------------------// LCD_set_xy//-----------------------------------------------------------------------------void LCD_set_xy( unsigned char x, unsigned char y ){unsigned char address;if (y == 0) address = 0x80 + x;elseaddress = 0xc0 + x;Write_COM(address);}//-----------------------------------------------------------------------------// LCD_write_string//-----------------------------------------------------------------------------void LCD_write_string(unsigned char n){unsigned char data1;for(n;n>0;n--){data1=*lcdpoint;writechar(data1);delaynms(100);lcdpoint++;delaynus(10);}}//-----------------------------------------------------------------------------// presskey//-----------------------------------------------------------------------------void presskey(void){unsigned char xdata DDCdata[16]={0x50,0x4c,0x45,0x41,0x53,0x45,0x20,0x50,0x52,0x45,0x53,0x53,0x20,0x4b,0x4 5,0x59};lcdpoint=&DDCdata;}//-----------------------------------------------------------------------------// lcd_key1//-----------------------------------------------------------------------------void lcd_key1(void){unsigned char xdata key1ok[6]={0x53,0x33,0x20,0x4f,0x4b,0x21};lcdpoint=&key1ok;}//-----------------------------------------------------------------------------// lcd_key2//-----------------------------------------------------------------------------void lcd_key2(void){unsigned char xdata key2ok[6]={0x53,0x34,0x20,0x4f,0x4b,0x21};lcdpoint=&key2ok;}//-----------------------------------------------------------------------------// lcd_key3//-----------------------------------------------------------------------------void lcd_key3(void){unsigned char xdata key3ok[6]={0x53,0x35,0x20,0x4f,0x4b,0x21};lcdpoint=&key3ok;}//-----------------------------------------------------------------------------// lcd_key4//-----------------------------------------------------------------------------void lcd_key4(void){unsigned char xdata key4ok[6]={0x53,0x36,0x20,0x4f,0x4b,0x21};lcdpoint=&key4ok;}//-----------------------------------------------------------------------------// delaynus//-----------------------------------------------------------------------------void delaynus(unsigned int q) //N us延时函数{for (i=0;i<q;i++){_nop_();}}本程序已经完全调试通过,欢迎参考。
系统辨识_6_多新息辨识理论与方法_丁锋

的最小二乘辨识算法或随机梯度等辨识算法有下列 形式: ^ ( t) = θ ^ ( t - 1 ) + L ( t ) e( t ) , θ e( t) : = 其中 L( t) ∈R 为算法增益向量( gain vector) , T ^ ( t - 1 ) ∈R 为标量新息 ( scalar innovay ( t) - φ ( t ) θ tion) , 即单新息( single innovation) . 这个算法可以这样描述: t 时刻的参数估计向量 ^ ( t) 是用增益向量 L ( t) 与标量新息 e ( t ) 的乘积, θ 对 ^ ( t - 1 ) 进行修正, ^ ( t) t - 1 时刻参数估计向量 θ 即θ ^ ( t - 1 ) 的基础上加上增益向量 L ( t ) 与新息 是在 θ e( t) 的乘积. 这种方法也称为新息修正辨识方法或 新息辨识方法. 上述算法中新息 e ( t ) 是标量, 我们把这个标量 in新息加以推广, 就导出了多新息辨识方法 ( multinovation identification method ) [24]. 多 新 息 辨 识 理 论 ( multiinnovation identification theory ) 就是将单新息 从新息修正角度提出多新息修 修正技术加以推广, 正技术辨识的概念, 建立多新息修正辨识方法, 简称 多新息辨识方法. 顾名思义, 多新息算法就是将新息加以推广. 对 将算法中的标量新息 e ( t ) ∈ R 推广 标量系统而言, t ) ∈ Rp , innova为新息向量 E ( p, 即 多 新 息 ( multin tion) , 为使矩阵乘法维数兼容, 增益向量 L ( t ) ∈ R t ) ∈R n × p , 须推广为增益矩阵( gain matrix) Γ( p, 那么 n
(完整版)电子信息与通信工程专业英语期末必考翻译

1。
"In most cases, these signals originate as sensory data from the real world:seismic vibrations visual images, sound waves, etc。
DSP isthe mathematics,the algorithms, and the techniques used to manipulate these signals after they have been converted into a digital form.”在大多数情况下,这些信号来源于人对真实世界的感觉,比如地震的震动,视觉图像,声音波形等。
数字信号处理是一种数学工具,是一种用来处理那些将上述信号转换成数字形式后的信号的算法和技术.2.Fourier’s representation of functionsas a superposition of sines and cosines has become Ubiquitous for both the analytic and numerical solution of differential equations and for the analysis and treatment of communication signals 函数的傅里叶表示,即将函数表示成正弦和余弦信号的叠加,这种方法已经广泛用于微分方程的解析法和数值法求解过程以及通信信号的分析和处理。
3。
If f (t ) is a nonperiodic signal, the summation of the periodic functions ,such as sine and cosine, does not accurately represent the signal. You couldartificially extend thesignal to make it periodicbut it would requireadditional continuity at the endpoints . 如果f(t)是非周期信号,那么用周期函数例如正弦和余弦的和,并不能精确的表示该信号f(t).你可以人为的拓展这个信号使其具有周期性,但是这要求在端点处附加连续性4。
振动分析和仪器的艺术达芬奇

Peak and RMS Comparison
Relationships of Acceleration, Velocity and Displacement
The Big Picture
Sensor(s)
Cables
Signal Conditioning
Data Acquisition & Storage
– Simple Harmonic Motion
• • Oscillation about a Reference Point Modeled Mathematically as…
x(t ) X sin t
Back to the Basics…
Period, T Unit Circle
RMS
• May result in amplifier output voltage becoming “Slew Rate Limited”
–
Sensor Cables
• •
–
Output of Sinusoid looks like this: What’s Happening?
The + part of the signal is being limited by the current available to drive the cable capacitance. In the – part of the sin wave, the op-amp must “sink” the current being discharged by the cable capacitance.
severity of reciprocal of Simple – •the Represented by vibratory Period motion •the Complex time delay CPS or two Hzas Pattern – •between Expressed Recognition • RPM •signals Peak to Peak Orders •Leading Zero to Peak – • RMS – •Lagging
DSP数字信号处理技术维基百科

Digital signal processingFrom Wikipedia, the free encyclopediaJump to: navigation, searchThis article needs additional citations for verification. Pleasehelp improve this article by adding citations to reliablesources. Unsourced material may be challenged and removed. (May2008)Digital signal processing (DSP) is the mathematical manipulation of an information signal to modify or improve it in some way. It is characterized by the representation of discrete time, discrete frequency, or other discrete domain signals by a sequence of numbers or symbols and the processing of these signals. Digital signal processing and analog signal processing are subfields of signal processing. DSP includes subfields like: audio and speech signal processing, sonar and radar signal processing, sensor array processing, spectral estimation, statistical signal processing, digital image processing, signal processing for communications, control of systems, biomedical signal processing, seismic data processing, etc.The goal of DSP is usually to measure, filter and/or compress continuous real-world analog signals. The first step is usually to convert the signal from an analog to a digital form, by sampling and then digitizing it using an analog-to-digital converter(ADC), which turns the analog signal into a stream of numbers. However, often, the required output signal is another analog output signal, which requires a digital-to-analog converter(DAC). Even if this process is more complex than analog processing and has a discrete value range, the application of computational power to digital signal processing allows for many advantages over analog processing in many applications, such as error detection and correction in transmission as well as data compression.[1]DSP algorithms have long been run on standard computers, on specialized processors called digital signal processor on purpose-built hardware such as application-specific integrated circuit (ASICs). Today there are additional technologies used for digital signal processing including more powerful general purpose microprocessors, field-programmable gate arrays (FPGAs), digital signal controllers (mostly for industrial apps such as motor control), and stream processors, among others.[2][edit] Signal samplingMain article: Sampling (signal processing)With the increasing use of computers the usage of and need for digital signal processing has increased. To use an analog signal on a computer, it must be digitized with an analog-to-digital converter. Sampling is usually carried out in two stages, discretization and quantization. In the discretization stage, the space of signals is partitioned into equivalence classes and quantization is carried out by replacing the signal with representative signal of the corresponding equivalence class. In the quantization stage the representative signal values are approximated by values from a finite set.The Nyquist–Shannon sampling theorem states that a signal can be exactly reconstructed from its samples if the sampling frequency is greater than twice the highest frequency of the signal; but requires an infinite number of samples. In practice, the sampling frequency is often significantly more than twice that required by the signal's limited bandwidth.[edit] DSP domainsIn DSP, engineers usually study digital signals in one of the following domains: time domain (one-dimensional signals), spatial domain (multidimensional signals), frequency domain, and wavelet domains. They choose the domain to process a signal in by making an informed guess (or by trying different possibilities) as to which domain best represents the essential characteristics of the signal. A sequence of samples from a measuring device produces a time or spatial domain representation, whereas a discrete Fourier transform produces the frequency domain information, that is the frequency spectrum. Autocorrelation is defined as the cross-correlation of the signal with itself over varying intervals of time or space.[edit] Time and space domainsMain article: Time domainThe most common processing approach in the time or space domain is enhancement of the input signal through a method called filtering. Digital filtering generally consists of some linear transformation of a numberof surrounding samples around the current sample of the input or output signal. There are various ways to characterize filters; for example:∙ A "linear" filter is a linear transformation of input samples; other filters are "non-linear". Linear filters satisfy the superposition condition, i.e. if an input is a weighted linear combination ofdifferent signals, the output is an equally weighted linearcombination of the corresponding output signals.∙ A "causal" filter uses only previous samples of the input or output signals; while a "non-causal" filter uses future input samples. A non-causal filter can usually be changed into a causal filter by adding a delay to it.∙ A "time-invariant" filter has constant properties over time; other filters such as adaptive filters change in time.∙ A "stable" filter produces an output that converges to a constant value with time, or remains bounded within a finite interval. An "unstable" filter can produce an output that grows without bounds, with bounded or even zero input.∙ A "finite impulse response" (FIR) filter uses only the input signals, while an "infinite impulse response" filter (IIR) uses both theinput signal and previous samples of the output signal. FIR filters are always stable, while IIR filters may be unstable.Filters can be represented by block diagrams, which can then be used to derive a sample processing algorithm to implement the filter with hardware instructions. A filter may also be described as a difference equation, a collection of zeroes and poles or, if it is an FIR filter, an impulse response or step response.The output of a linear digital filter to any given input may be calculated by convolving the input signal with the impulse response.[edit] Frequency domainMain article: Frequency domainSignals are converted from time or space domain to the frequency domain usually through the Fourier transform. The Fourier transform converts the signal information to a magnitude and phase component of each frequency.Often the Fourier transform is converted to the power spectrum, which is the magnitude of each frequency component squared.The most common purpose for analysis of signals in the frequency domain is analysis of signal properties. The engineer can study the spectrum to determine which frequencies are present in the input signal and which are missing.In addition to frequency information, phase information is often needed. This can be obtained from the Fourier transform. With some applications, how the phase varies with frequency can be a significant consideration.Filtering, particularly in non-realtime work can also be achieved by converting to the frequency domain, applying the filter and then converting back to the time domain. This is a fast, O(n log n) operation, and can give essentially any filter shape including excellent approximations to brickwall filters.There are some commonly used frequency domain transformations. For example, the cepstrum converts a signal to the frequency domain through Fourier transform, takes the logarithm, then applies another Fourier transform. This emphasizes the frequency components with smaller magnitude while retaining the order of magnitudes of frequency components.Frequency domain analysis is also called spectrum-or spectral analysis. [edit] Z-plane analysisMain article: Z-transformWhereas analog filters are usually analysed in terms of transfer functions in the s plane using Laplace transforms, digital filters are analysed in the z plane in terms of Z-transforms. A digital filter may be described in the z plane by its characteristic collection of zeroes and poles. The z plane provides a means for mapping digital frequency (samples/second)to real and imaginary z components, where for continuous periodicsignals and ( is the digital frequency). This is useful for providing a visualization of the frequency response of a digital system or signal.[edit] WaveletMain article: Discrete wavelet transformAn example of the 2D discrete wavelet transform that is used in JPEG2000. The original image is high-pass filtered, yielding the three large images, each describing local changes in brightness (details) in the original image. It is then low-pass filtered and downscaled, yielding an approximation image; this image is high-pass filtered to produce the three smaller detail images, and low-pass filtered to produce the final approximation image in the upper-left.In numerical analysis and functional analysis, a discrete wavelet transform (DWT) is any wavelet transform for which the wavelets are discretely sampled. As with other wavelet transforms, a key advantage it has over Fourier transforms is temporal resolution: it captures both frequency and location information (location in time).[edit] ApplicationsThe main applications of DSP are audio signal processing, audio compression, digital image processing, video compression, speech processing, speech recognition, digital communications, RADAR, SONAR, seismology and biomedicine. Specific examples are speech compression and transmission in digital mobile phones, room correction of sound in hi-fi and sound reinforcement applications, weather forecasting, economic forecasting, seismic data processing, analysis and control of industrial processes, medical imaging such as CAT scans and MRI, MP3 compression, computer graphics, image manipulation, hi-fi loudspeaker crossovers and equalization, and audio effects for use with electric guitar amplifiers.[edit] ImplementationDepending on the requirements of the application, digital signal processing tasks can be implemented on general purpose computers (e.g. supercomputers, mainframe computers, or personal computers) or with embedded processors that may or may not include specialized microprocessors called digital signal processors.Often when the processing requirement is not real-time, processing is economically done with an existing general-purpose computer and the signal data (either input or output) exists in data files. This is essentially no different than any other data processing, except DSP mathematical techniques (such as the FFT) are used, and the sampled data is usually assumed to be uniformly sampled in time or space. For example: processing digital photographs with software such as Photoshop.However, when the application requirement is real-time, DSP is often implemented using specialised microprocessors such as the DSP56000, the TMS320, or the SHARC. These often process data using fixed-point arithmetic, though some more powerful versions use floating point arithmetic. For faster applications FPGAs[3] might be used. Beginning in 2007, multicore implementations of DSPs have started to emerge from companies including Freescale and Stream Processors, Inc. For faster applications with vast usage, ASICs might be designed specifically. For slow applications, a traditional slower processor such as a microcontroller may be adequate. Also a growing number of DSP applications are now being implemented on Embedded Systems using powerful PCs with a Multi-core processor。
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∗
This author is supported in part by the Tateishi Science and Technology Foundation.
1
shift and gain change. This property is directly connected to the physical interpretation of the steady-state response, and is particularly effective in identification and analysis of oscillatory nature of systems. The basic objective in sampled-data systems is to control continuous-time plants via discrete-time controllers. Therefore, it is more natural to evaluate its performance in continuous-time, and this has been indeed the driving force for the recent interest in the study of sampled-data systems with built-in intersample behavior : [6], [5], [8], [9], [3], [15], [14], [17], [19], just to name a few. In this respect, and particularly so for H ∞ type design, it is more desirable to evaluate the system performance in terms of the frequency response of the system in the continuous-time behavior. There is, however, a conceptual difficulty. If we take intersample behavior into account, the state transition is time-varying, and the notion of steady-state response against sinusoidal inputs is not directly at hand. Suppose that we apply a sinusoid sin ωt to a sampled-data system Σ with sampling period h. If we sample sin ωt, the resulting signal at sampled instants is {sin ωkh}∞ k =0 , and this is not even periodic unless ω and h synchronize (i.e., kωh = 2nπ for some n, k ). Therefore, the notion of steady-state response against sinusoidal inputs is nontrivial to justify, and so is the frequency response. Recently, two approaches have been proposed to overcome this difficulty. One is based upon the so-called lifting technique [20, 21], which is to view sampled-data systems as a time-invariant system by extending the input/output spaces to function spaces. In this way, the time-varying nature of sampled-data systems is absorbed into parameters of operators acting upon function spaces, so that the resulting system may be viewed time-invariant, thereby allowing for such time-invariant notions as transfer functions. The other approach [1, 2] utilizes the impulse modulation formula and uses the fact that if a sinusoidal input ejωt is applied, then its corresponding output of the sampled-data system is expressed as, instead of that proportional to the original ejωt , an infinite sum of its sideband components yn exp(j (ω +2nπ/h)t). With this expansion, an 2 -type induced norm is introduced. What is left open is their mutual relationship. This paper establishes the relationships between the two approaches and proves that the notions of frequency responses defined by the two methods are identical. This also gives rise to a useful relation among various system concepts and techniques. On one hand, it gives rise to the interpretation of alias effects in the frequency response in the first method: it appears as the totality of all sideband components related to the chosen particular angular frequency. This is perhaps better understood in the second method. On the other hand, the present study exhibits a clearer interpretation of the relationship of the second method to Fourier series, and establishes a direct connection with the H ∞ norm. The paper is organized as follows: Section 2 introduces the basic model via 2
1
Introduction
The notion of frequency response plays a crucial role in evaluating the performance of time-invariant continuous-time systems. This is based upon the fundamental property that if we apply a sinusoidal input to a time-invariant stable continuoustime system, then the output is also a sinusoid with the same frequency with phase
lifting, and transfer operators. Section 3 gives two independent notions of frequency responses and their gain functions, and establishes their equivalence. Section 4 derives some concrete formulae for the frequency response operator, which have been derived via impulse modulation formulae, for two fundamental cases. NOTATION and CONVENTION The notation is quite standard. The sampling period is denoted by h. L2 [0, h] and L2 [0, ∞) are the spaces of Lebesgue square integrable functions on [0, h] and [0, ∞), respectively. In general, we omit superscripts to denote the dimension of the range spaces. So we write simply L2 [0, h] instead of (L2 [0, h])n , etc. Likewise, 2 is the space of (vector-valued) square summable sequences. When we need to explicitly show the space X where such sequences take n their values on, we write 2 X instead. For a vector x ∈ IR , its Euclidean norm will be denoted by |x|, to make the distinction clear from the L2 norm. In contrast, if we write ϕ , it will usually denote an L2 (or 2 ) norm, or the operator norm induced by it. C [0, h] denotes the space of continuous functions on [0, h]. Laplace and z -transforms are denoted by L[ϕ](s) and Z [ϕ](z ), respectively. When no confusion can arise, we may also write ϕ ˆ(s), ϕ ˆ(z ), depending on the context.