《通信与电子信息科技英语》课件unit 18.ppt

1、Abstract1 iterative adj.迭代的iterationn.迭代 noniterative algorithm 非迭代算法2 tap n.抽头3 superiority n.优越(性),优势(to,over)4 factorization n.因式分解 spectral factorization 谱(因式)分解5 derivation n.推导6 circumvent vt.防止，避免7 convergence n.集中,收敛8 recursive adj.递归的 9 transversal adj.横向的transversal equalizer横向均衡器10 corrob

2、orate vt.确证；证实11 descent n.降下,降落 steepest descent method 最陡下降法12 orthogonalize vt.使成为正交,使正交化13 gradient n.梯度gradient algorithm梯度算法Abstract14 sequentially adv.继续地,从而15 bipolar adj.双极的16 robust adj.健壮的17 inphase adj.同相（位）的18 quadrature n.正交,90相移 quadrature phase 正交(相)位移,90相位移19 misalignment n.失调20 coh

3、erent adj.相干的noncoherentadj.非相干的21 concurrent adj.并行的,并发的22 bulk n.大小23 limited n.有限公司24 quasi adj.准的，类似的 quasisynchronous adj.准同步的25 drastically adv.急剧地,彻底地26 mitigation n.缓和，减轻mitigation technique缓和技术Abstract27 simulation n.仿真，模拟28 commence vt.开始；着手29 telescope n.望远镜30 gap n.间隙31 dedicated adj.专用的

4、32 dish n.碟形卫星天线33 astronomical adj.天文学的34 maser n.微波激射器 microwave amplification by stimulated emission of radiation35 pulsar n.脉冲星36 continuum n.连续体；连续性continuum source连续源37 prototype n.样机,原型38 transponder n.转发器satellite transponder卫星转发器39 watermark n.水印Abstract40 imperceptible adj.感觉不到的(to)41 ceps

5、trum n.倒谱cepstrum domain例谱域42 cepstral adj.倒谱的43 hyperbolic adj.双曲线的44 tangent n.正切 hyperbolic tangent function 双曲正切函数45 attenuation n.衰减rain attenuation雨衰,雨致衰减46 radiometer n.辐射计47 domestic adj.国产的,国内的 domestic communication satellite 国内通讯卫星，本国通讯卫48 countermeasure n.对策,反对手段49 forthcoming adj.即将来临的5

6、0 meteorology n.气象学radiometeorologyn.无线电气象学51 tropical adj.热带的 52 modification n.改进Abstract1 impulse response 冲激响应2 intersymbol interference 码间干扰,符号间干扰3 MLSE（maximum likelihood sequence estimator)最大似然序列估计器4 computational efficiency 计算效率5 adaptive equalizer 自适应均衡器6 IIR(infinite impulse response)无限冲激响

7、应7 Wiener filter 维纳滤波器8 system identification 系统识别9 in common with 和一样10 LMS(least mean squares)最小均方11 in parallel with 与并行,与同时12 prior knowledge 先验知识Abstract13 delay line 延迟线14 consistent with 与一致15 convergence rate(rate of convergence)收敛速度16 correction term 修正项17 correlation matrix 相关矩阵,相关系数矩阵18 o

8、rder of magnitude 数量级19 MSE(meansquare error)均方误差20 LSI(large scale integration)大规模集成化21 computing element 计算元件22 be competitive with 可与竞争23 full digital 全数字式24 FM(frequency modulation)调频制,调频Abstract25 IF(intermediate frequency)中频26 AGC(automatic gain control 自动增益控制27 AFC(automatic frequency control

9、)自动频率控制28 DRS(Data Relay System)数据中继系统29 LEO(low Earth orbit)近地轨道30 ESA(European Space Agency)欧洲航天局31 Doppler shift 多普勒频移，多普勒频偏32 under contract 受合同的约束33 band limitation 频带限制34 selfnoise(self noise)固有噪声,自身噪声 35 CSIRO（Commonwealth Scientific and Industrial Research Organization）(澳大利亚)联邦科学与工业研究组织36 of

10、fline 离线的,脱机的Abstract37 GPS（Global Positioning System）全球定位系统38 DFE（decision feedback equalizer)判决反馈均衡器39 a soft decision 软判决40 hard limiter 硬限幅器41 interfering signal 干扰信号42 compact disk (只读)光盘43 local minima 局部极小,局部最小值44 atmospheric absorption 大气吸收45 to date 到目前为止，迄今46 under way 进行中,在行进47 millimeter

11、 wave 毫米波Abstract A new noniterative algorithm is proposed to estimate the sampled impulse response of unknown channels.With the proper choice of the training sequence,implementation of the proposed channel estimator requires only additions and subtractions,i.e.,no multiplications or divisions are n

12、eeded.Moreover,the structure of the estimator is so simple that it can be easily implemented using an ordinary microprocessor with minimal storage.The channel estimate can serve in reducing the effects of intersymbol interference either by the maximum likelihood sequence estimator(MLSE)using the Vit

13、erbi algorithm,or by channel equalization with a direct solution to obtain the optimum equalizer taps.A new procedure for the latter case is proposed here using the LevinsonTrench algorithm for fast startup of adaptive equalizers in noisy environments.The performance of the proposed algorithm is eva

14、luated Abstractthrough simulation and it is compared to some of the existing techniques.Computational efficiency is also taken into account.Results of simulation show the superiority of the proposed scheme.The development of an adaptive infinite impulse response(IIR)linear equalizeris described.Usin

15、g discrete time Wiener filtering theory,a closed form for the optimum meansquare error IIR filter is derived.A performance comparison using both minimum and nonminimum phase channels indicates the complexity/performance advantages inherent in the IIR system compared to an optimum finite impulse resp

16、onse(FIR)solution.The minimum phase spectral factorization,which is an integral part of the derivation of the IIR equalizer,may be circumvented through the use of a Kalman equalizer such as that originally proposed by Lawrence and Kaufman.The structure is made adaptive by using a system identificati

17、on algorithm operating in parallel with a Kalman Abstractequalizer.In common with Luvison and Pirani,a least mean squares(LMS)algorithm was chosen for the system identification because the input to the channel is white and hence the LMS algorithm will produce consistent predictable results with litt

18、le added complexity.A new technique is introduced which both estimates the variance of channel noise and compensates the Kalman filter for errors in the estimate of the channel impulse response.Computer simulation results show that the convergence performance of this new adaptive IIR filter is rough

19、ly equivalent to an FIR equalizer which is trained using a recursive least squares algorithm.However,the order of the new filter is always lower than the FIR filter.It is shown how a Kalman filter may be applied to the problem of setting the tap gains of transversal equalizers to minimize meansquare

20、 distortion.In the presence of noise and without prior knowledge about Abstractthe channel,the filter algorithm leads to faster convergence than other methods,its speed of convergence depending only on the number of taps.Theoretical results are given and computer simulation is used to corroborate th

21、e theory and to compare the algorithm with the classical steepest descent method.A comparison is made of several selforthogonalizing adjustment algorithms for linear tapped delay line equalizers.Exercises.Please translate the following words and phrases into Chinese.1.MLSE 2.IIR 3.rain attenuation 4

22、.LMS 5.LSI 6.system identification 7.IF 8.AFC 最大似然序列估计器最大似然序列估计器(maximum likelihood sequence estimator)无限冲激响应无限冲激响应(infinite impulse response)雨致衰减雨致衰减 最小均方最小均方(least mean squares)大规模集成化大规模集成化(largescale integration)系统识别系统识别中频中频(intermediate frequency)自动频率控制自动频率控制(automatic frequency control)Exercise

23、s9.reference antenna 10.ESA 11.cepstrum domain 12.DFE13.AGC14.data relay facility15.Maser16.spectral factorization17.LEO18.MSE参考天线参考天线 欧洲航天局欧洲航天局(European Space Agency)倒谱域倒谱域判决反馈均衡器判决反馈均衡器(decision feedback equalizer)自动增益控制自动增益控制(automatic gain control)数据中继设备数据中继设备微波激射器微波激射器(microwave amplification

24、by stimulated emission of radiation)谱谱(因式因式)分解分解 近地轨道近地轨道(Low Earth Orbit)均方误差均方误差(Mean Square Error)Exercises.Please translate the following words and phrases into English.1.卫星转发器2.自适应均衡器 3.维纳滤波器 4.局部最小值5.多普勒频移 6.全数字调制解调器 7.全球定位系统 8.双曲正切函数satellite transponderadaptive equalizerWiener filter local m

25、inima Doppler shiftfull digital modem Global Positioning System hyperbolic tangent functionExercises9.音频信号10.准同步的 11.数量级12.频带限制13.硬限幅器14.毫米波15.收敛速度16.计算效率17.先验知识18.相关矩阵audio signalquasisynchronous order of magnitude band limitation hard limiter millimeter wave convergence rate(rate of convergence)co

26、mputational efficiency prior knowledgecorrelation matrixExercises.Fill in the blanks with the missing word(s).1.The channel estimate can serve in reducing the effects of inter symbol Interference either by the maximum likelihood sequence estimator(MLSE)using the Viterbi algorithm,or by channel equal

27、ization with a direct solution to obtain the optimum equalizer taps.2.The performance of the proposed algorithm is evaluated through simulation andit is compared to some of the exiting techniques.Computational efficiency is also taken into account.3.In common with Luvision and Pirani,least mean squa

28、res(LMS)algorithm was chosen for the system identification because the input to the channel is white and hence the LMS algorithm will produce consistent predictable results with littleadded complexity.4.This paper shows how a Kalman filter may be applied to the problem of setting the tap gains of tr

29、ansversal equalizers to minimize mean square distortion.Exercises5.In the presence of noise and without prior knowledge about the channel,the filter algorithm leads to faster convergence than other methods.6.The direct solution is based on the algorithms devised by Levinson and Trench.7.With the ava

30、ilability of large scale integration(LSI)bipolar computing elements,these algorithms are competitive with iterative procedures.8.It is shown that the delay can be adjusted to achieve minimum MSE with respect to that parameters.9.An all digital demodulator/detector is presented which is suitable for

31、both analog FM and digital phase/frequency modulations.10.The spread spectrum receiver and modem is presently under development by Satellites International Limited under contract to the European Space Agency.Exercises11.Band limitation is achieved with no detection loss by means of Nyquist chip shap

32、ing,leading to a simple all digital demodulator structure.12.Impairments due to satellite transponder distortions are also evaluated.13.The full digital modem structure is presented together with possible applications to mobile and VSAT satellite communications.14.Simulations can be useful,but are n

33、ot a substitute for real data.15.We insert a watermark into the cepstral components of the audio signal.16.The first experiments involving Kaband earthspace propagation measurements in Japan were carried out in 1977.17.In Japan at present,advanced propagation studies are under way in wideband digita

34、ltransmission satellite systems for the forthcoming multimedia era.Exercises18.A novel noniterative algorithm is (were,be,is,was)proposed to estimate (estimating,estimate)the sampled impulse response of unknown channels.19.In this paper we propose a new algorithm for digital audio Watermarking in the cepstrum domain.20.In particular the impact of the use of a kind of network clock Synchronization on the overall MCD complexity is investigated in detail.