电子信息专业英语09

单击此处编辑母版标题样式,单击此处编辑母版文本样式,第二级,*,信息科学与电子工程专业英语,单击此处编辑母版标题样式,单击此处编辑母版文本样式,第二级,*,Technical English,For Information Science and Electronic Engineering,Unit 9,Electronics: Digital Signals and Signal Processing,Part I,Digital Signal Processing,3,New Words,sub-field,分领域，子领域,sonar,声呐,sensor array,传感器阵列,biomedical,生物特征的,algorithm,算法,abbreviate,缩写，缩略,purpose-designed,针对目的设计的,application-specific,面向应用的,integrated circuits,集成电路,autocorrelation,自相关,wavelet,小波,baseband,基带,informed,有知识的，有见闻的,spatial domain,空间域,cross-correlation,互相关,interval,间隔,discretization,离散化,quantization,量化,partition,分割，分区,finite set,有限的集,4,New Words,theorem,定理,scenario,情节，方案,carrier,载波,ingredient,成分，因素,demodulation,解调,enhancement,增强,filtering,滤波,weighted,加权的,characterize,描绘，刻画性质,superposition,叠加,causal,因果关系的,converge,收敛,diverge,发散,bounded,有界的,transfer functions,传递函数,block diagram,方框图,derive,推导,magnitude,大小,cepstrum,倒谱,logarithm,对数,5,New Words,FIR (finite impulseresponse filter),有限冲击响应滤波器,IIR (infinite impulse response filter),无限冲击响应滤波器,field-programmablegate array (FPGA),现场可编程门阵列,computer aidedtomography (CAT),计算机断层扫描,seismology,地震学,equalization,均衡,hifi (high fidelity),高保真（音乐）,reinforcement,加强,animation,动画,loudspeaker,扬声器,floating point,浮点,arithmetic,算术,magnetic resonanceimaging (MRI),磁共振成像,fixed-point arithmetic,定点运算，整数运算,6,1,Digital signal processing (DSP) is the study of signals in a digital representation and the processing methods of these signals. DSP and analog signal processing are sub-fields of signal processing. DSP includes sub-fields like,audio and speech signal processing, sonar and radar signal processing, sensor array processing, spectral estimation, statistical signal processing, image processing, signal processing for communications, biomedical signal processing, etc.,音频及语音信号处理、声纳和雷达信号处理、传感器阵列处理、谱估计、统计信号处理、图像处理、通信信号处理、生物医学信号处理,7,2,Since the goal of DSP is usually to measure or filter continuous real-world analog signals, the first step is usually to convert the signal from an analog to a digital form, by using an analog to digital converter. Often, the required output signal is another analog output signal, which requires a digital to analog converter.,数字信号处理的目标通常是测量连续的真实世界的模拟信号或对其滤波,8,3,The algorithms required for DSP are sometimes performed using specialized computers, which make use of specialized microprocessors called digital signal processors (also abbreviated DSP).,These process signals in real time, and are generally purpose-designed application-specific integrated circuits (ASICs).,1,这些数字信号处理器实时处理信号，通常是针对具体目的而设计的专用集成电路ASIC。,9,3,When flexibility and rapid development are more important than unit costs at high volume, DSP algorithms may also be implemented using field-programmable gate arrays (FPGAs).,2,当灵活性和快速开发比大批量生产的本钱更重要时，DSP算法也可以用现场可编程门阵列来实现。,10,4 DSP domains,In DSP, engineers usually study digital signals in one of the following domains: time domain (one-dimensional signals), spatial domain (multidimensional signals), frequency domain, autocorrelation domain, and wavelet domains.,They choose the domain in which to process a signal by making an informed guess (or by trying different possibilities) as to which domain best represents the essential characteristics of the signal.,3,他们按某些依据来猜测或试验不同的可能性那一个域能够最好地表示信号的本质特性来选择在其中进展信号处理的域。,11,4,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.,4,Autocorrelation is defined as the cross-correlation of the signal with itself over varying intervals of time or space.,从测量设备得到的样本序列产生信号的时域或空域表示，而离散Fourier变换那么产生频域表示即频谱。,12,5 Signal sampling,With the increasing use of computers the usage and need of digital signal processing has increased. In order to use an analog signal on a computer it must be digitized with an analog to digital converter (ADC).,Sampling is usually carried out in two stages, discretization and quantization,.,采样通常分两步实现：离散化和量化,13,5,In the discretization stage, the space of signals is partitioned into equivalence classes and discretization is carried out by replacing the signal with representative signal of the corresponding equivalence class.,5,In the quantization stage,the representative signal values are approximated by values from a finite set,.,在离散化阶段，信号空间被分割为相等的区间，用相应区间的代表性信号值代替信号本身。,用有限集中的值来近似代表性的信号值,14,6,In order for a sampled analog signal to be exactly reconstructed, the Nyquist-Shannon sampling theorem must be satisfied.,This theorem states that the sampling frequency must be greater than twice the bandwidth of the signal,.,定理规定：采样频率必须大于两倍的信号带宽,15,6,In practice, the sampling frequency is often significantly more than twice the required bandwidth. The most common bandwidth scenarios are: DCBW (“baseband); and fcBW, a frequency band centered on a carrier frequency (“direct demodulation).,采样频率通常远大于信号带宽的两倍,16,7,A digital to analog converter (DAC) is used to convert the digital signal back to analog. The use of a digital computer is a key ingredient into digital control systems.,17,8 Time and space domains,The most common processing approach in the time or space domain is enhancement of the input signal through a method called filtering.,Filtering generally consists of some transformation of a number of surrounding samples around the current sample of the input or output signal,. There are various ways to characterize filters; for example:,滤波通常由在输入或输出信号当前样本周围的许多样本的某种变换组成,18,8,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 of different signals, the output is an equally weighted linear combination of the corresponding output signals.6,线性滤波器满足叠加条件，就是说，如果输入是不同信号的加权线性组合，输出就是各信号相应输出的同样加权线性组合。,19,8,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.,“时不变滤波器对时间具有不变的性质，诸如自适应滤波器等其它滤波器随时间而改变,20,8,Some filters are “stable, others are “unstable. A stable filter produces an output that converges to a constant value with time, or remains bounded within a finite interval. An unstable filter produces output which diverges.,稳定的滤波器产生的输出随时间收敛于一个不变的值，或在有限的时间间隔内保持有界,21,8,A “finite impulse response (FIR) filter uses only the input signal, while an “infinite impulse response filter (IIR) uses both the input signal and previous samples of the output signal. FIR filters are always stable, while IIR filters may be unstable.,而无限脉冲响应IIR滤波器同时使用输入信号和以前的输出信号样本,22,9,Most filters can be described in Z-domain (a superset of the frequency domain) by their transfer functions.,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.,7,滤波器也可以用差分方程或一组零极点表示，对于,FIR,滤波器还可以用冲击响应或阶跃响应表示。,23,9,The output of an FIR filter to any given input may be calculated by convolving the input signal with the impulse response.,Filters can also be represented by block diagrams which can then be used to derive a sample processing algorithm to implement the filter using hardware instructions,.,滤波器还可以用构造图来表示，它能用来推导样本处理算法，以便使用硬件指令实现滤波器,24,10 Frequency domain,Signals 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.,Fourier,变换将信号信息变换成每个频率的幅度和相位成分,25,11,The most common purpose for analysis of signals in the frequency domain is analysis of signal properties. The engineer can study the spectrum to get information of which frequencies are present in the input signal and which are missing.,26,12,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, and then applies another Fourier transform. This emphasizes the frequency components with smaller magnitude while retaining the order of magnitudes of frequency components.,8,例如倒谱用,Fourier,变换将信号转换到频域，取对数，然后再作第二次,Fourier,变换。这就强调了幅度较小的频率成分同时保持了频率分量的数量级。,27,13 Applications,The 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.,28,13,Specific examples are speech compression and transmission in digital mobile phones, room matching equalization of sound in HiFi and sound reinforcement applications, weather forecasting, economic forecasting, seismic data processing, analysis and control of industrial processes, computer-generated animations in movies, medical imaging such as CAT scans and MRI, image manipulation, high fidelity loudspeaker crossovers and equalization, and audio effects for use with electric guitar amplifiers.,29,14 Implementation,Digital signal processing is often implemented using specialized microprocessors such as the MC56000 and the TMS320.,These often process data using fixed-point arithmetic, although some versions are available which use floating point arithmetic and are more powerful,.,它们通常使用定点算法处理数据，尽管也有一些使用浮点算法，运算能力更强大,30,14,For faster applications FPGAs might be used. Beginning in 2007, multicore implementations of DSPs have started to emerge. For faster applications with vast usage, ASICs might be designed specifically. For slow applications, a traditional slower processor such as a microcontroller can cope.,Part II,General Concepts of Digital Signal Processing,32,New Words,special-purpose,专用,terminology,术语,sinusoidal,正弦的,context,上下文，背景,casually,随便地,rigid,坚硬的，刚性的,pursue,追求，从事,excitation,激励,differential equation,微分方程,difference equation,差分方程,time-invariant,时不变的,lumped system,集总系统,increment,增量,presampling,预采样,waveform,波形,strategy,策略,multiplex,复用,dynamic range,动态范围,compatible,兼容,subtraction,减法,33,New Words,multiplication,乘法,reconstruction,重建,holding circuit,保持电路,extrapolate,外推,interpolate,内插,curve-fitting,曲线拟合,prescribed,预定的,intervene,插入，干预,intervening,期间的,bandlimited,限带的,aliasing,混叠,erroneous,错误的,wagon,四轮马车,spoke,轮辐,spurious,假的，伪造的,arbitrarily,任意地,uncertainty,不确定性,quantum,量子，量化,demltiplexer,解复用器,telemetry,遥测,34,1,There have been tremendous demands in the use of digital computers and special-purpose digital circuitry for performing varied signal processing functions that were originally achieved with analog equipment. The continued evolution of inexpensive integrated circuits has led to a variety of microcomputers and minicomputers that can be used for various signal processing functions.,35,1,It is now possible to build special-purpose digital processors within much smaller size and lower cost constraints of systems previously all analog in nature.,1,现在有可能在比以往全模拟系统小得多，而且本钱也低得多的限制下构成专用数字处理器。,36,2,We will provide a general discussion of the basic concepts associated with digital signal processing. To do so, it is appropriate to discuss some common terms and assumptions.,Wherever possible, the definitions and terminology will be established in accordance with the recommendations of the IEEE Group on Audio and Electroacoustics,.,可能之处，定义和术语将依照,IEEE,音频和电声小组的推荐,37,3,An analog signal is a function that is defined over a continuous range of time and in which the amplitude may assume a continuous range of values,. Common examples are the sinusoidal function, the step function, the output of a microphone, etc. The term,analog,apparently originated from the field of analog computation, in which voltages and currents are used to represent physical variables, but it has been extended in usage.,模拟信号是定义在连续时间上的函数，其幅度取值是连续的,38,4,Continuous-time signal is a function that is defined over a continuous range of time, but in which the amplitude may either have a continuous range of values or a finite number of possible values,. In this context, an analog signal could be considered as a special case of a continuous-time signal. In practice, however, the terms,analog,and,continuous-time,are interchanged casually in usage and are often used to mean the same thing.,连续时间信号是定义在连续时间上的函数，但是它的幅度可能是连续值也可以是有限的可能值,39,4,Because of the association of the term,analog,with physical analogies, preference has been established for the term,continuous-time,. Nevertheless, there will be cases in which the term,analog,will be used for clarity, particularly where it relates to the term,digital,.,2,由于“模拟一词与物理类比的关联，已经确立了优先使用“连续时间这一术语。不过有时为了清楚起见也用“模拟一词，特别是与“数字相联系时。,40,5,The term quantization describes the process of representing a variable by a set of distinct values,. A quantized variable is one that may assume only distinct values,.,经量化的变量只能取离散值,41,6,A discrete-time signal is a function that is defined only at a particular set of values of time,. This means that the independent variable, time, is quantized. If the amplitude of a discrete-time signal is permitted to assume a continuous range of values, the function is said to be a sampled-data signal. A sampled-data signal could arise from sampling an analog signal at discrete values of time.,离散时间信号是定义在某些特定时间值上的函数,42,7,A digital signal is a function in which both time and amplitude are quantized. A digital signal may always be represented by a sequence of numbers in which each number has a finite number of digits.,43,8,The terms,discrete-time,and,digital,are often interchanged in practice and are often used to mean the same thing.,A great deal of the theory underlying discrete-time signals is applicable to purely digital signals, so it is not always necessary to make rigid distinctions,. The term,discrete-time,will more often be used in pursuing theoretical developments, and the term,digital,will more often be used in describing hardware or software realizations.,许多基于离散时间信号的定理适用于纯数字信号，因此没有必要总是对两者作严格的区分,44,9,A system can be described by any of the preceding terms according to the type of hardware or software employed and the type of signals present,. Thus, reference can be made to,analog systems,continuous-time systems,discrete-time systems,digital systems, etc.,根据使用的硬件或软件的类型和当前信号的类型，一个系统可以用任意的前述术语来描述,45,10,A linear system is one in which the principle of superposition applies.,A linear system can be described by linear differential or difference equations,. A time-invariant linear system is one in which the parameters are fixed and do not vary with time.,线性系统可以用线性的微分或差分方程来描述,46,11,A lumped system is one that is composed of finite nonzero elements satisfying ordinary differential or difference equation relationships, as opposed to a distributed system, satisfying partial differential equation relationships.,3,集总系统是由有限非零元素构成，满足常微分或差分方程的系统，与满足偏微分方程的分布式系统相对应。,47,12,The standard form for numerical processing of a digital signal is the binary number system. The binary number system makes use only of the values 0 and 1 to represent all possible numbers.,48,12,The number of levels,m,that can be represented by a number having,n,binary digits (bits) is given by,Conversely, if,m,is the number of possible levels required, the number of bits required is the smallest integer greater than or equal to log2,m,.,4,可用n位二进制n比特表示的等级数m由m= 2n给出。反过来，如果m是要求的等级数，所需的比特数是大于等于log2 m的最小整数。,m,= 2,n,49,13,The process by which digital signal processing is achieved will be illustrated by a simplified system in which the signal is assumed to vary from 0 to 7 volts and in which 8 possible levels (at 1 V increments) are used for the binary numbers.,5,实现数字处理的过程将用一个简化系统来说明，假定信号在,0V,到,7V,之间变化，以,1V,为增量，用,8,种可能的值表示成二进制数。,50,13,A block diagram is shown in Figure 9.1, and some waveforms of interest are shown in Figure 9.2. The signal is first passed through a continuous-time presampling filter whose function will be discussed later.,Pre-sampling filter,Sampler and A/D converter,Digital signal processor,D/A converter and filters,Analog,signal in,Analog,signal,out,000,010,011,111,111,110,100,011,001,8,T,0,T,2,T,3,T,4,T,5,T,6,T,7,T,t,8,T,0,T,2,T,3,T,4,T,5,T,6,T,7,T,t,x,(,t,) and,quantized samples,51,13,The signal is then read at intervals of,T,seconds by a sampler. These samples must then be quantized to one of the standard levels. Although there are different strategies employed in the quantization process, one common approach, which will be assumed here, is that a sample is assigned to the nearest level. Thus, a sample of value 4.2 V would be quantized to 4 V, and a sample of value 4.6 V would be quantized to 5 V.,52,14,This process for the signal given is illustrated in Figure 9.2. The pulses representing the signal have been made very narrow to illustrate the fact that other signals may be inserted, or multiplexed, in the empty space. These pulses may then be represented as binary numbers.,In order that these numbers could be seen on the figure, each has been shown over much of the space in a given interval.,为了使这些数字可以从图中看到，每组都显示在给定间隔的空档处,53,14,In practice, if other signals are to be inserted, the pulses representing the bits of the binary numbers could be made very short.,A given binary number could then be read in a very short interval at the beginning of a sampling period, thus leaving most of the time available for other signals,.,一个给定的二进制数就可以在采样周期开场的很短间隔内读到，这样就给其它信号留出了大局部的可用时间,54,15,The process by which an analog sample is quantized and converted to a binary number is called analog-to-digital (A/D) conversion. In general,the dynamic range of the signal must be compatible with that of the A/D converter employed, and the number of bits employed must be sufficient for the required accuracy.,信号的动态范围要和所用的A/D转换器相一致，为了到达所要求的准确度，要使用足够的比特数,55,16,The signal can now be processed by the type of unit appropriate for the application intended. This unit may be a general-purpose computer or minicomputer, or it may be a special unit designed specifically for this purpose. At any rate, it is composed of some combination of standard digital circuits capable of performing the various arithmetic functions of addition, subtraction, multiplication, etc. In addition, it has logic and storage capability.,56,17,At the output of the processor, the digital signal can be converted to analog form again. This is achieved by the process of digital-to-analog (D/A) conversion. In this step, the binary numbers are first successively converted back to continuous-time pulses. The,gaps,between the pulses are then filled in by a reconstruction filter.,57,17,This filter may consist of a holding circuit, which is a special circuit designed to hold the value of a pulse between successive sample values.,In some cases, the holding circuit may be designed to interpolate the output signal between successive points according to some prescribed curve-fitting strategy.,6,In addition to a holding circuit, a basic continuous-time filter may be employed to provide additional smoothing between points.,在某些情况下，可设计保持电路，将输出信号在连续样点之间按照设定的曲线拟合方法进展内插。,58,18,A fundamental question that may arise is whether or not some information has been lost in the process. After all, the signal has been sampled only at discrete intervals of time; is there something that might be missed in the intervening time intervals Furthermore, in the process of quantization, the actual amplitude is replaced by the nearest standard level, which means that there is a possible error in amplitude.,信号仅仅在离散的时间间隔处被采样；在介于时间间隔内是否有一些信息丧失了呢,59,19,In regard to the sampling question, it will be shown that, if the signal is bandlimited, and if the sampling rate is greater than or equal to twice the highest frequency, the signal can theoretically be recovered from its discrete samples.,7,关于采