编辑推荐： 中国教育部最近通知，部属各重点科研基地须有自己的定期学术刊物，刊物的形式必须是正式杂志或有固定网址的电子期刊。 《语言学论丛》（26辑起）是教育部重点科研基地北京大学汉语语言学研究中心的定期学术文集，由商务印书馆每年出版两辑。根据上述文件精神，《语言学论丛》将同设电子期刊。自26辑以后的论文，在出版一年之后，免费发布在“北大学术期刊网”（http：／／e－journals．pku．edu．cn／Jour／Index．aspx？JIID＝21）上。发布的内容为：文章的中英文标题、提要、关键词；作者、资助来源、作者单位、通信处；参考文献；文章的第一章或相当于引言的段落。 电子期刊的设立，相信有利于论文进入相关的学术论文检索系统，有利于学术评估和交流。 目录： 从隐含到呈现（上）——试论中古词汇的一个本质变化 方言接触对变调语法的影响 论“一音一义” 《中原音韵》的原始著作权和它的基础方言问题 上古音构拟的检验标准问题 普通话上声强重音的声学表现 上海话声调右扩展的语音实质 小句焦点的韵律结构类型及其变化 古汉语中宾语的表层隐现条件及其解释 佛教文献中“何”系疑问代词的兴替演变 陕北晋语沿河方言时制系统研究 被动标记“让”在当代汉语中的发展 羡余否定的类别、成因与功用 说“安”字 ABSTRACTS（英文提要）

《30天搞定英语四级》系列丛书是一部为非英语专业（公共英语）的英语学习者快速高分挑战“全国大学英语四级考试”量身定制的自主复习用书。丛书由七个分册组成，分别是备考词汇分册；基础知识分册；听力分册；阅读分册；语法与词汇分册；完形填空·翻译·简答分册以及写作分册。丛书的主创人员为多年从事四、六级英语教学的教师，他们教学经验丰富，备考指导有方，其门下弟子历年的四级通过率几近百分之一百。本丛书还吸纳了个别四、六级攻关高手参加部分章节的编写工作，他们成功的备考经历为丛书提供了独到的见解，也将为广大意欲奋战四级沙场的莘莘学子奉献宝贵的经验。本书为《30天搞定英语四级》系列丛书之一，即标准四级考试试题的第四部分。1996年之前，这部分的题型一直是“完形填空”（Cloze）。1996年1月和7月，“翻译”（Translation）取代“完形填空”；1997年1月，出现了“简答题”（ShortAnswerQuestion，简称SAQ）。自此，第四部分的考题就成了四级试卷中最变幻莫测的题型。本书为《30天搞定英语四级》系列丛书之完形填空·翻译·简答分册，具有以下特点：一、真正的30天突破本书的30天突破是科学合理地以一定的题量划分的，每一天均有各种题型的特训练习，有助于学生在较短时间内有序、全面、有重点地抓住重点，有助于合理有效地安排时间，逐步掌握四级新题型，从而彻底征服大学英语四级。二、科学性与针对性本书编委严格遵循《大学英语四级考试大纲》的出题准则，仔细研究并分析历年之考试真题，大胆预测来年可能出现的试题形式，在编写过程中尽量做到承前启后，既有科学性，又有针对性。我们精心选择内容新颖、题材丰富、语言地道的英语文章，以科学的态度用心设计模拟试题，用诙谐幽默的语气详尽讲解解题方略，使考生在没有教师指导的情况下，能轻松自主复习，成功过级。三、极富人性化的服务体系本书根据《艾宾浩斯遗忘曲线》为读者安排了本书的学习计划，只要读者严格遵循复习，将确保学习效果（详见使用说明）。在书末，我们都为读者准备了学习心得记录页，方便读者记录每天的学习心得和难点重点，读者在学完本书后可以将其剪下装订成册，方便复习。本书由以下三部分组成：最经典的题型——完形填空：作者列表统计分析了1995年以来“完形填空”的出现次数和题型分布情况，研究、剖析往年之考试真题，从宏观全文布局到微观词、句、结构各方面为考生详尽讲解、献计支招。“挑战真题”部分提供了5套20篇习题，文章全部取自CNN（有线新闻电视网）、NewYorkTimes（纽约时报）、VOA（美国之音）等知名媒体及英语学习网站，材料新颖，语言地道，难度适宜；习题后附有答案与解析，内容详实，非常有说服力。最实在的题型——翻译：作者从“英译汉”障碍剖析入手，仔细分析阻碍准确翻译的各种因素，联系实例仔细介绍大学英语四级考试“英译汉”的标准以及攻克“英译汉”的步骤和技巧。20篇文章组成的“挑战真题”练习，经过广泛而细致的挑选，内容丰富，文体多样。文章辅以具有针对性的翻译例题和翔实的解析，译文地道准确，极具参考价值。最热门的题型——简答题：作者介绍了四级考试的简答题评分标准与原则。通过分析最近两次的试卷的命题形式，预测今后可能出现的考试题型，并推荐行之有效的答题策略。20篇习题，为广大考生提供了高信度的复习资料；答案讲解循循善诱，使学生在自主复习时，如同有专业又耐心的教师亲临指导一般。本书为备考四级的自主复习用书，读者可以自行随意使用。本书编委愿奉献以下建议供大家参考：仔细阅读每一种题型中的解题指导部分，了解该题型的出题原则、评分标准以及在解答该题型过程中可能存在的障碍；分析领会作者的计谋招数。尝试本书作者为您策划的技巧计谋，练习挑战真题中的各种题型；根据参考答案与解析仔细校对答案。您可以自行设定时间，一次做一套练习（oneunit），或一次做一篇练习（onepassage）。根据四级考试的标准，第四部分考题的给定时间是15分钟，因此，您应遵循这一原则，视情况安排好做题时间。参考数据为15分钟/篇。您千万不要自作主张放弃某种题型的复习。考试题型每年都在变化，最变化多端的就是新题型部分。所以，为了顺利过级，考出优异成绩，我们要知己知彼全面准备。

英语大突破系列教材。全书以语法为主线，每章由“典型真题”、“语法精要”、“疑难点悟”、“翻译实践”、“参考译文”五部分组成。

本书注重联系大纲并结合高等英语教学特点，对历年实考真题命题特点与趋势进行介绍，并收录2000至2003年的英语应用能力考试B级试题且进行解答。

认知语言学是20世纪80年代以来在西方兴起的新的语言学流派。《认知语言学与隐喻研究》分七章，首先介绍了认知语言学的理论体系及研究现状；随后运用认知语言学的观点和研究方法，对英汉诗歌中的隐喻现象进行分析。全书文笔流畅，作者结合实例，将认知语言学的理论知识讲解得深入浅出，既可为一般读者提供认知语言学的入门指导，也可为相关领域的研究者提供最新的研究成果。

本书将理论与实践完美地结合起来，通过具有启发性的解释和精心挑选的例子，采用读者可以理解的数学表示方法，对数字信号处理 （DSP） 技术进行了解释，帮助读者从整体上掌握DSP，并逐步掌握较高层次的DSP概念和应用。本书第1版就很畅销，可读性强，受到专业人员的广泛喜爱。第2版以此为基础，除包括周期采样、离散傅里叶变换 （DFT） 、快速傅里叶变换 （FFT） 、数字滤波器、离散希尔伯特变换、采样速率转换、量化、信号取均值等内容外，还扩展了无限脉冲响应滤波器 （IIR） 、脉冲频率采样和内插式有限脉冲响应滤波器 （FIR） 等内容，增添了多种新的速率系统，包括多相滤波器和串级积分器-梳状有限脉冲响应滤波器 （FIR） 。本书是工程师以及专业技术人员掌握和应用DSP技术的最佳参考书。本书的主要内容·包含实际的、直观的、丰富的DSP例子。·强调实际的、日常的DSP应用和解决方案。·提供了全新的正交信号处理内容，包括易于理解的三维空间图。·包括即使是经验丰富的专业人士也可能忽略的技术方法。·涵盖了频率采样、内插式FIR、CIC等重要滤波器。·提供了流行的数字信号处理技巧。Richard G. Lyons 毕业于美国加州阿克隆大学， 获得BSEE学位. 他是美国加州Mount View市Besser公司的咨询系统工程师和讲师， 为许多著名公司举办过DSP方面的技术培训， 这些公司包括：摩托罗拉. 洛克希德·马丁. 霍尼韦尔. 德州仪器. 诺思洛普·格鲁曼. 朗迅. 诺基亚. 高通. 国家半导体. 通用动力. 西门子 （现为英飞灵） 等. 他做过国家安全局和TRW公司的众多DSP系统的首席硬件工程师， 负责系统设计. 开发. 测试和安装. Lyons曾在加利福尼亚大学圣克鲁兹分校任教， 并发表过很多有关DSP的文章. 作为IEEE Signal Processing杂志的副主编， 他创建和编辑了"DSP Tips & Tricks"专栏.This book is an expansion of the original Understanding Digital Signal Process-ing textbook published in 1997 and， like the first edition， its goal is to help be-ginners understand this relatively new technology of digital signal processing （DSP）. Additions to this second edition include：· Expansion and clarification of selected spectrum analysis and digital ill-tering topics covered in the first edition making that material more valu-able to the DSP beginner.· Expanded coverage of quadrature （complex I/Q） signals. In many cases we used three-dimension time and frequency plots to enhance the de-scription of， and give physical meaning to， these two-dimensional sig-nals.· With the new emphasis on quadrature signals， material was added de-scribing the Hilbert transform and how it‘s used in practice to generate quadrature signals.· Discussions of Frequency Sampling， Interpolated FIR， and CIC filters;giving these important filters greater exposure than they‘ve received in past DSP textbooks.· A significant expansion of the popular "Digital Signal Processing Tricks" chapter.· Revision of the terminology making it more consistent with the modern day language of DSP. It‘s traditional at this point in the preface of a DSP textbook for the au-thor to tell readers why they should learn DSP. I don‘t need to tell you how important DSP is in our modern engineering world， you already know that.I‘ll just say that the future of electronics is DSP， and with this book you will not be left behind. LEARNING DIGITAL SIGNAL PROCESSING Learning the fundamentals， and how to speak the language， of digital signal processing does not require profound analytical skills or an extensive background in mathematics. All you need is a little experience with elementary al-gebra， knowledge of what a sinewave is， this book， and enthusiasm. This may sound hard to believe， particularly if you‘ve just flipped through the pages of this book and seen figures and equations that look rather complicated. The content here， you say， looks suspiciously like the material in technical journals and textbooks that， in the past， have successfully resisted your attempts to understand. Well， this is not just another book on digital signal processing. This book‘s goal is to gently provide explanation followed by illustration， not so that you may understand the material， but that you must understand the material？ Remember the first time you saw two people playingchess？ The game probably appeared to be mysterious and confusing. As younow know， no individual chess move is complicated. Given a little patience， the various chess moves are easy to learn. The game‘s complexity comes from deciding what combinations of moves to make and when to make them. So it is with understanding digital signal processing. First we learn the fundamental rules and processes， and then practice using them in combination. If learning digital signal processing is so easy， then why does the subject have the reputation of being hard to understand？ The answer lies partially in how the material is typically presented in the literature. It‘s difficult to convey technical information， with its mathematical subtleties， in written form. It‘s one thing to write equations， but it‘s another matter altogether to explain what those equations really mean from a practical standpoint， and that‘s the goal of this book. Too often， written explanation of digital signal processing theory appears in one of two forms： either mathematical miracles occur and the reader is simply given a short and sweet equation without further explanation， or the reader is engulfed in a flood of complex variable equations and phrases such as "it is obvious that，" and "with judicious application of the homogeneity property." In their defense， authors usually do provide the needed information， but too often the reader must figuratively grab a pick and shovel， put on a miner‘s helmet， and try to dig the information out of a mountain of mathematical expressions. （This book presents the results of several fruitful mining expeditions.） How many times have you followed the derivation of an equation， after which the author states they‘re going to illustrate that equation with an examplewhich turns out to be just another equation？ Although mathematics is necessary to describe digital signal processing， I‘ve tried to avoid overwhelming the reader with math because a recipe for technical writing that‘s too rich in equations is hard for the beginner to digest. The intent of this book is expressed by a popular quote from E.B. White in the introduction of his Elements of Style （Macmillan Publishing， New York， 1959）： "Will （Strunk） felt that the reader was in serious trouble most of the time， a man floundering in a swamp， and that it was the duty of anyone attempting to write English to drain the swamp quickly and get his man up on dry ground， or at least throw him a rope." I‘ve attempted to avoid the traditional instructor-student relationship， but rather to make reading this book like talking to a friend while walking in the park. I‘ve used just enough mathematics to develop a fundamental understanding of the theory， and then illustrate that theory with practical examples. THE JOURNEY Learning digital signal processing is not something you accomplish; it‘s a journey you take. When you gain an understanding of some topic， questions arise that cause you to investigate some other facet of digital signal processing. Armed with more knowledge， you‘re likely to begin exploring further aspects of digital signal processing much like those shown in the following diagram. This book is your tour guide during the first steps of your journey. You don‘t need a computer to learn the material in this book， but it would sure help. DSP simulation software allows the beginner to verify signal processing theory through the time-tested trial and error process. In particular software routines that plot signal data， perform the fast Fourier transforms， and analyze digital filters would be very useful. As you go through the material in this book， don‘t be discouraged if your understanding comes slowly. As the Greek mathematician Menaechmus curtly remarked to Alexander the Great， when asked for a quick explanation of mathematics， "There is no royal road to mathematics." Menaechmus， was confident in telling Alexander the only way to learn mathematics is through careful study. The same applies to digital signal processing. Also， don‘t worry if you have to read some of the material twice. While the concepts in thisbook are not as complicated as quantum physics， as mysterious as the lyricsof the song Louie Louie， or as puzzling as the assembly instructions of a metal shed， they do get a little involved. They deserve your attention and thought. So go slow and read the material twice if you have to; you‘ll be glad you did. If you show persistence， to quote a phrase from Susan B. Anthony， "Failure is impossible." COMING ATTRACTIONS Chapter 1 begins by establishing the notation used throughout the remainder of the book. In that chapter we introduce the concept of discrete signal sequences， show how they relate to continuous signals， and illustrate how those sequences can be depicted in both the time and frequency domains. In addition， Chapter 1 defines the operational symbols we‘ll use to build our signal processing system block diagrams. We conclude that chapter with a brief introduction to the idea of linear systems and see why linearity enables us to use a number of powerful mathematical tools in our analysis. Chapter 2 introduces the most frequently misunderstood process in digital signal processing， periodic sampling. Although it‘s straightforward to grasp the concept of sampling a continuous signal， there are mathematical subtleties in the process that require thoughtful attention. Beginning gradually with simple examples of low-pass sampling， and progressing to the interesting subject of bandpass sampling， Chapter 2 explains and quantifies the frequencydomain ambiguity （aliasing） associated with these important topics. Chapter 3 is devoted to one of the foremost topics in digital signal processing， the discrete Fourier transform （DFT） used for spectrum analysis. Coverage begins with detailed examples illustrating the important properties of the DFT and how to interpret DFT spectral results， progresses to the topic of windows used to reduce DFT leakage， and discusses the processing gain afforded by the DFT. The chapter concludes with a detailed discussion of the various forms of the transform of rectangular functions that the beginner is likely to encounter in the literature. That last topic is included there to clarify and illustrate the DFT of both real and complex sinusoids. Chapter 4 covers the innovation that made the most profound impact on the field of digital signal processing， the fast Fourier transform （FFT）. There we show the relationship of the popular radix-2 FFT to the DFT， quantify the powerful processing advantages gained by using the FFT， demonstrate why the FFT functions as it does， and present various FFT implementation structures. Chapter 4 also includes a list of recommendations to help the reader use the FFT in practice. Chapter 5 ushers in the subject of digital filtering. Beginning with a simple low-pass finite impulse response （FIR） filter example， we carefully progress through the analysis of that filter‘s frequency domain magnitudeand phase response. Next we learn how window functions affect， and can be used to design， FIR filters. The methods for converting low-pass FIR filter designs to bandpass and high pass digital filters are presented， and the popular Remez Exchange （Parks McClellan） FIR filter design technique is intro duced and illustrated by example. In that chapter we acquaint the reader with， and take the mystery out of， the process called convolution. Proceeding through several simple convolution examples， we conclude Chapter 5 with a discussion of the powerful Convolution Theorem and show why it‘s so useful as a qualitative tool in understanding digital signal processing. Chapter 6 is devoted to a second class of digital filters， infinite impulse response （IIR） filters. In discussing several methods for the design of IIR filters， the reader is introduced to the powerful digital signal processing analysis tool called the z-transform. Because the z-transform is so closely related to the continuous Laplace transform， Chapter 6 starts by gently guiding the reader from the origin， through the properties， and on to the utility of the Laplace transform in preparation for learning the z-transform. We‘ll see how IIR filters are designed and implemented， and why their performance is so different from FIR filters. To indicate under what conditions these filters should be used， that chapter concludes with a qualitative comparison of the key properties of FIR and IIR filters. Chapter 7 introduces two specialized digital filter types that have not re- ceived their deserved exposure in traditional DSP textbooks. Called frequency sampling and interpolated FIR filters， and providing enhanced lowpass filtering computational efficiency， they belong in our arsenal of filter design techniques. Although these are FIR filters， their introduction is delayed to this chapter because familiarity with the z-transform （in Chapter 6） makes the properties of these filters easier to understand. Chapter 8 presents a detailed description of quadrature signals （also called complex signals）. Because quadrature signal theory has become so important in recent years， in both signal analysis and digital communication implementations， it deserves its own chapter. Using three-dimensional illustrations， this chapter gives solid physical meaning to the mathematical notation， processing advantages， and use of quadrature signals. Special emphasis is given to quadrature sampling （also called complex down-conversion）. Chapter 9 provides a mathematically gentle， but technically thorough， description of the Hilbert transform：a process used to generate a quadrature （complex） signal from a real signal. In this chapter we describe the properties， behavior， and design of practical Hilbert transformers. Chapter 10 presents a brief introduction to the fascinating， and very use- ful， process of sample rate conversion （changing the effective sample rate of discrete data sequences through decimation or interpolation）. Sample rate conversion：so useful in improving the performance and reducing the computational complexity of many signal processing operations：is essentially an exercise in lowpass filter design. As such， polyphase and cascaded integratorcomb filters are also described in this chapter. Chapter 11 covers the important topic of signal averaging. There we learn how averaging increases the accuracy of signal measurement schemes by reducing measurement background noise. This accuracy enhancement is called processing gain， and that chapter shows how to predict the processing gain associated with averaging signals in both the time and frequency domains. In addition， the key differences between coherent and incoherent averaging techniques are explained and demonstrated with examples. To complete that chapter the popular scheme known as exponential averaging is covered in some detail. Chapter 12 presents an introduction to the various binary number formats the reader is likely to encounter in modern digital signal processing. We establish the precision and dynamic range afforded by these formats along with the inherent pitfalls associated with their use. Our exploration of the critical subject of binary data word width （in bits） naturally leads us to a discussion of the numerical resolution limitations of analog to digital （A/D） converters and how to determine the optimum A/D converter word size for a given application. The problems of data value overflow roundoff errors are covered along with a statistical introduction to the two most popular remedies for overflow， truncation， and rounding. We end that chapter by covering the interesting subject of floating point binary formats that allow us to overcome most of the limitations induced by fixed point binary formats， particularly in reducing the iii effects of data overflow. Chapter 13 provides a collection of tricks of the trade used to make digital signal processing algorithms more efficient. Those techniques are compiled into a chapter at the end of the book for two reasons. First， it seems wise to keep our collection of tricks in one chapter so that we‘ll know where to find them in the future. Second， many of these schemes require an understanding of the material from the previous chapters， so the last chapter is an appropriate place to keep our arsenal of clever tricks. Exploring these techniques in detail verifies and reiterates many of the important ideas covered in previous chapters. The appendices include a number of topics to help the beginner under- stand the nature and mathematics of digital signal processing. A comprehensive description of the arithmetic of complex numbers is covered in Appendix A， while Appendix B derives the often used， but seldom explained， closed form of a geometric series. The subtle aspects and two forms of time reversal in discrete systems （of which zero-phase digital filtering is an application） are explained in Appendix C. The statistical concepts of mean， variance， and standard deviation are introduced and illustrated in Appendix D， while Appendix E provides a discussion of the origin and utility of the logarithmic decibel scale used to improve the magnitude resolution of spectral representations. Appendix F， in a slightly different vein， provides a glossary of the ter-minology used in the field of digital filters. Preface iii 1 DISCRETE SEQUENCES AND SYSTEMS 1.1Discrete Sequences and Their Notation 2 1.2Signal Amplitude， Magnitude， Power 8 1.3Signal Processing Operational Symbols 9 1.4Introduction to Discrete Linear Time-Invariant Systems12 1.5Discrete Linear Systems 12 1.6Time-Invariant Systems 17 1.7The Commutative Property of Linear Time-Invariant Systems 18 1.8Analyzing Linear Time-Invariant Systems 19 2 PERIODIC SAMPLING 21 2.1Aliasing： Signal Ambiquity in the Frequency Domain 21 2.2Sampling Low-Pass Signals 26 2.3Sampling Bandpass Signals 30 2.4Spectral Inversion in Bandpass Sampling 39 3 THE DISCRETE FOURIER TRANSFORM45 3.1Understanding the DFT Equation 46 3.2DFT Symmetry 58 3.3DFT Linearity 60 3.4DFT Magnitudes 61 3.5DFT Frequency Axis 62 3.6DFT ShiftingTheorem 63 3.7Inverse DFT 65 3.8DFT Leakage 66 3.9Windows 74 3.10DFT Scalloping Loss 82 3.11DFT Resolution， Zero Padding， and Frequency-Domain Sampling 83 3.12DFT Processing Gain 88 3.13The DFT of Rectangular Functions 91 3.14The DFT Frequency Response to a Complex Input 112 3.15The DFT Frequency Response to a Real Cosine Input 116 3.16The DFT Single-Bin Frequency Response to a Real Cosine Input 117 3.17Interpreting the DFT 120 4 THE FAST FOURIER TRANSFORM125 4.1Relationship of the FFT to the DFT 126 4.2Hints on Using FFTs in Practice 127 4.3FFT Software Programs 131 4.4Derivation of the Radix-2 FFT Algorithm 132 4.5FFT Input/Output Data Index Bit Reversal 139 4.6Radix-2 FFT Butterfly Structures 141 5 FINITE IMPULSE RESPONSE FILTERS 151 5.1An Introduction to Finite Impulse Response FIR Filters 152 5.2Convolution in FIR Filters 157 5.3Low-Pass FIR Filter Design 167 5.4Bandpass FIR Filter Design 183 5.5Highpass FIR Filter Design 184 5.6Remez Exchange FIR Filter Design Method 186 5.7Half-Band FIR Filters 188 5.8Phase Response of FIR Filters 190 5.9A Generic Description of Discrete Convolution 195 6 INFINITE IMPULSE RESPONSE FILTERS 211 6.1An Introduction to Infinite Impulse Response Filters 212 6.2The Laplace Transform 215 6.3The z-Transform 228 6.4Impulse Invariance IIR Filter Design Method 243 6.5Bilinear Transform IIR Filter Design Method 259 6.6Optimized IIR Filter Design Method 270 6.7Pitfalls in Building IIR Digital Filters 272 6.8Improving IIR Filters with Cascaded Structures 274 6.9A Brief Comparison of IIR and FIR Filters 279 SPECIALIZED LOWPASS FIR FILTERS283 7.1Frequency Sampling Filters： The Lost Art 284 7.2Interpolated Lowpass FIR Filters 319 8 QUADRATURE SIGNALS335 8.1Why Care About Quadrature Signals 336 8.2The Notation of Complex Numbers 336 8.3Representing Real Signals Using Complex Phasors 342 8.4A Few Thoughts on Negative Frequency 346 8.5Quadrature Signals in the Frequency Domain 347 8.6Bandpass Quadrature Signals in the Frequency Domain 350 8.7Complex Down-Conversion 352 8.8A Complex Down-Conversion Example 354 8.9An Alternate Down-Conversion Method 358 9 THE DISCRETE HILBERTTRANSFORM361 9.1Hilbert Transform Definition 362 9.2Why Care About the Hilbert Transform？ 364 9.3Impulse Response of a Hilbert Transformer 369 9.4Designing a Discrete Hilbert Transformer 371 9.5Time-Domain Analytic Signal Generation 377 9.6Comparing Analytical Signal Generation Methods 379 10 SAMPLE RATE CONVERSION381 10.1Decimation 382 10.2Interpolation 387 10.3Combining Decimation and Interpolation 389 10.4Polyphase Filters 391 10.5Cascaded Integrator-Comb Filters 397 11 SIGNAL AVERAGING411 11.1Coherent Averaging 412 11.2Incoherent Averaging 419 11.3Averaging Multiple Fast Fourier Transforms 422 11.4Filtering Aspects of Time-Domain Averaging 430 11.5Exponential Averaging 432 12 DIGITAL DATA FORMATS AND THEIR EFFECTS 439 12.1Fixed-Point Binary Formats 439 12.2Binary Number Precision and Dynamic Range 445 12.3Effects of Finite Fixed-Point Binary Word Length 446 12.4Floating-Point Binary Formats 462 12.5Block Floating-Point Binary Format 468 13 DIGITAL SIGNAL PROCESSING TRICKS 471 13.1Frequency Translation without Multiplication 471 13.2High-Speed Vector-Magnitude Approximation 479 13.3Frequency-Domain Windowing 484 13.4Fast Multiplication of Complex Numbers 487 13.5Efficiently Performing the FFT of Real Sequences 488 13.6Computing the Inverse FFT Using the Forward FFT 500 13.7Simplified FIR Filter Structure 503 13.8Reducing A/D Converter Quantization Noise 503 13.9A/D Converter Testing Techniques 510 13.10Fast FIR Filtering Using the FFT 515 13.11Generating Normally Distributed Random Data 516 13.12Zero-Phase Filtering 518 13.13Sharpened FIR Filters 519 13.14Interpolating a Bandpass Signal 521 13.15Spectral Peak Location Algorithm 523 13.16Computing FFT Twiddle Factors 525 13.17Single Tone Detection 528 13.18The Sliding DFT 532 13.19The Zoom FFT 541 13.20A Practical Spectrum Analyzer 544 13.21An Efficient Arctangent Approximation 547 13.22Frequency Demodulation Algorithms 549 13.23DC Removal 552 13.24Improving Traditional CIC Filters 556 13.25Smoothing Impulsive Noise 561 13.26Efficient Polynomial Evaluation 563 13.27Designing Very High-Order FIR Filters 564 13.28Time-Domain Interpolation Using the FFT 568 13.29Frequency Translation Using Decimation 571 13.30Automatic Gain Control （AGC） 571 13.31Approximate Envelope Detection 574 13.32A Quadrature Oscillator 576 13.33Dual-Mode Averaging 578 APPENDIX A. THE ARITHMETIC OF COMPLEX NUMBERS 585 A.1Graphical Representation of Real and Complex Numbers 585 A.2Arithmetic Representation of Complex Numbers 586 A.3Arithmetic Operations of Complex Numbers 588 A.4Some Practical Implications of Using Complex Numbers 593 APPENDIX B. CLOSED FORM OF A GEOMETRIC SERIES595 APPENDIX C. TIME REVERSAL AND THE DFT599 APPENDIX D. MEAN， VARIANCE， AND STANDARD DEVIATION603 D.1Statistical Measures 603 D.2Standard Deviation， or RMS， of a Continuous Sinewave 606 D.3The Mean and Variance of Random Functions 607 D.4The Normal Probability Density Function 610 APPENDIX E. DECIBELS （DB AND DBM） 613 E.1Using Logarithms to Determine Relative Signal Power 613 E.2Some Useful Decibel Numbers 617 E.3Absolute Power Using Decibels 619 APPENDIX F. DIGITAL FILTER TERMINOLOGY621 APPENDIX G. FREQUENCY SAMPLING FILTER DERIVATIONS633 G.1Frequency Response of a Comb Filter 633 G.2Single Complex FSF Frequency Response 634 G.3Multisection Complex FSF Phase 635 G.4Multisection Complex FSF Frequency Response 636 G.5Real FSF Transfer Function 638 G.6Type-IV FSF Frequency Response 640 APPENDIX H. FREQUENCY SAMPLING FILTER DESIGN TABLES643 INDEX 657 ABOUT THE AUTHOR667

语言在交际中规范，因为语言是在交际之中的。有人不同意这个看法，所以对待规范的态度、原则和方法就不同。从说话了解人不容易，由于有语言学，有很多语言学的分支。不从说话了解人，又很难同这个人、这一类人说话。要同这个人、这类人把话说好，就是语言交际好。语言交际好的目的和原则是寻求进一步的合作。不能合作、不想合作就连话也不说了，说话，就是还有合作和寻求进一步合作的可能。语言是人为了合作而出现、为了合作得好而发展的。本书试图对语言规范化的理论和实践作些比较系统的探讨。全书内容大体可以分为三个部分。第一部分（前三章）比较系统地梳理半个世纪里我们在语言规范化过程中的得与失，分析语言应用和语言规范的演变特征及其存在的问题。目的是“以史为鉴”，重在建设。第二部分（中间三章）主要是探讨一些理论问题，这些问题都是语言规范化中比较根本的问题。语言规范化的出发点和目的地是什么，什么样的原则才是语言规范化的根本原则，有了这个基础才能更好地认识语言调节和语言变化的关系。第三部分（后三章）则是在前面理论探讨的基础上来具体考察一些长期引起争议的而又折射出很多深层次问题的新现象，并在实践中丰富理论。这里并不试图从社会发展、文化等外部因素进行探讨，而是探讨这些现象产生的语言内部原因和生成的可能性问题，主要是从语义的角度来分析背后的一些东西。也许从语言结构本身认识得比较清楚了，才能更好地回答对这些语言现象的批评，才能使得出的结论更具操作性。<

《30天搞定英语四级》系列丛书是一部为非英语专业（公共英语）的英语学习者快速高分挑战“全国大学英语四级考试”量身定制的自主复习用书。丛书由七个分册组成，分别是备考词汇分册；基础知识分册；听力分册；阅读分册；语法与词汇分册；完形填空·翻译·简答分册以及写作分册。丛书的主创人员为多年从事四、六级英语教学的教师，他们教学经验丰富，备？贾傅加蟹剑涿畔碌茏永甑乃募锻ü始附俜种话佟１敬允榛刮闪烁霰鹚摹⒘豆ス馗呤植渭硬糠终陆诘谋嘈垂ぷ鳎浅晒Φ谋缚季允樘峁┝硕赖降募猓步愦笠庥苷剿募渡吵〉妮份费ё臃钕妆蟮木椤？本书为《30天搞定英语四级》繁殖丛书之写作分册，编写紧扣大学英语四级作文考试的具体要求和评分标准，充分考虑学生的实际水平及需要，突出了以下几个特点。一、真正的30天突破本书的30天突破是科学合理地以一定的题量划分的，每一天均有各种题型的特训练习，有助于学生在较短时间内有序、全面、有重点地抓住重点，有助于合理有效地安排时间，逐步掌握四级新题型，从而彻底征服大学英语四级。二、系统全面本分册对甲级作文中要求掌握的所有文体，包括论述文、应用文、图表作文和观察描述类作文以及目前还未现但大纲要求掌握的关键词作文，进行了全面详尽的介绍，并结合历年实考作文及部分学生习作进行了深入浅出的讲解，对各类文体和种类、结构、特点及考试题型和评分标准等做了透彻的剖析，帮助学生从宏观上把握四级写作考试的要求和特点。三、注重实用书中所有的内容都围绕着“实用”两字展开，不仅提供了审题构思、布局谋篇的方法，而且还结合实例，指明学生在文章连贯性、遣词造句等方面的薄弱环节，分析作文获得高分的要素，并提供了学生可以套用的句型。在抛砖引玉篇中还提供了20个模拟题及范文，供学生练习和模仿。四、选材真实新颖本分册中选用的解析材料大都来自近年来四级考试的原始评分样卷和学生平时的作文，未做任何改动。读者可参参照这些文章，客观评价自己的水平，并对如何避免和修改类似的文章做到心中有数，同时可以从范文中汲取精华，达到提高自己写作水平，在考试中？竦酶叻值哪康摹？五、极富人性化的服务体系本书根据《艾宾浩斯遗忘曲线》为读者安排了本书的学习计划，只要读者严格遵循复习，将确保学习效果（详见使用说明）。在书末，我们都为读者准备了学习心得记录页，方便读者记录每天的学习心得和难点重点，读者在学完本书后可以将其剪下装订成册，方便复习。全书分为“知己知彼”、“步步为“羸”（营）”、“抛砖引玉”天个部分。在“知己知彼篇中”，读者可以了解四级作文的选题与评分标准以及学生作文中常见的错误。“步步为羸”篇从审题构思，布局谋篇着手，一步一步教会学生如何运用写作技巧，提高写作水平。此外，这个部分还提供了许多大家可以直接套用的句型，为读者找到了一条提高写作水平的“捷径”。“抛砖引玉篇”中提供了20个模拟题及其思路分析和范文，为读者更好地进行甲级写作特训。

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