數(shù)字信號處理

內(nèi)容概要

　　為了給讀者在理論和實踐應用之間進行合理的平衡，本書嚴謹?shù)亟榻B了現(xiàn)代數(shù)字信號處理的基本概念和技術(shù)，并介紹了相關(guān)的算法和應用。本書涵蓋了線性離散時間系統(tǒng)分析的時域和頻域方法，還涉及了諸如采樣、數(shù)字濾波器設計、濾波器實現(xiàn)、去卷積、插值、狀態(tài)矢量空間方法、頻譜分析等相關(guān)主題的內(nèi)容。本書不僅要求對諸多示例、練習的理解，而且更強調(diào)對數(shù)字信號算法進行軟件實現(xiàn)的實踐環(huán)節(jié)。
　　本書特點：
　　·覆蓋離散傅立葉變換（DFT）和快速傅立葉變換（FFT）算法，并對其進行了更加合理清晰的重組——介紹DFT，并在闡明傅立葉分析后描述其快速計算
　　·描述模擬信號模數(shù)轉(zhuǎn)換中涉及的運算和技術(shù)
　　·在時域研究線性時不變離散時間系統(tǒng)和離散時間信號的特性
　　·考慮雙邊z變換和單邊z變換，并描述了求z反變換的方法
　　·在頻域分析信號與系統(tǒng)，給出連續(xù)時間信號與離散時間信號的傅立葉級數(shù)與傅立葉變換
　　·實現(xiàn)無限沖激響應（IIR）與有限沖激響應（FIR）系統(tǒng)的結(jié)構(gòu)形式，包括直接型、級聯(lián)型、并聯(lián)型、格型和格梯型
　　·采樣頻率轉(zhuǎn)換基礎(chǔ)與多采樣率轉(zhuǎn)換系統(tǒng)
　　·功率譜估計的詳細測試，并討論了非參數(shù)方法、基于模型的方法和基于特征分解的方法，包括MUSIC算法和ESPRIT算法
　　·全書囊括了許多實例，并提供大約500個可解決的問題
　　本書既適合作為本科生學習離散系統(tǒng)和數(shù)字信號處理課程的教材，又適合研究生一年級學習數(shù)字信號處理課程時作為教材使用。

作者簡介

　　John
G.Proakis長期擔任美國東北大學的電氣工程教授，并擔任該校電氣與計算機工程系主任之職達14年之久。他分別從麻省理工學院和哈佛大學獲得了碩士和博士學位。Proakis教授是眾多成功教材的作者，其教材在世界上具有相當?shù)挠绊懥Α?/pre>


書籍目錄
PREFACE
1 INTRODUCTION
1.1 Signals,Systems,and Signal Processing
1.1.1 Basic Elements of a Digital Signal Processing System
1.1.2 Advantages of Digital over Analog Signal Processing
1.2 Classification of Signals
1.2.1 Multichannel and Multidimensional Signals
1.2.2 Continuous-Time Versus Discrete-Time Signals
1.2.3 Continuous-Valued Versus Discrete-Valued Signals
1.2.4 Deterministic Versus Random Signals
1.3 The Concept of Frequency in Continuous-Time and Discrete-Time
Signals
1.3.1 Continuous-Time Sinusoidal Signals
1.3.2 Discrete-Time Sinusoidal Signals
1.3.3 Harmonically Related Complex Exponentials
1.4 Analog-to-Digital and Digital-to-Analog Conversion
1.4.1 Sampling of Analog Signals
1.4.2 The Sampling Theorem
1.4.3 Quantization of Continuous-Amplitude Signals
1.4.4 Quantization of Sinusoidal Signals
1.4.5 Coding of Quantized Samples
1.4.6 Digital-to-Analog Conversion
1.4.7 Analysis of Digital Signals and Systems Versus Discrete-Time
Signals and Systems
1.5 Summary and References
Problems
2 DISCRETE-TIME SIGNALS AND SYSTEMS
2.1 Discrete-Time Signals
……
2.2 Discrete-Time Systems
2.3 Analysis of Discrete-Time Linear Time-Invariant Systems
2.4 Discrete-Time Systems Described by Difference Equations
2.5 Implementation of Discrete-Time Systems
2.6 Correlation of Discrete-Time Signals
2.7 Summary andReferences
Problems
3 THE Z-TRANSFORM AND ITS APPLICATION TO THEANALYSIS OF
LTlSYSTEMS
3.1 The z-Transform
3.2 Properties o fthe z-Transform
3.3 Rational z-Transforms
3.4 Inversion of the z-Transform
3.5 The One-sided z-Transform
3.6 Analysis of Linear Time-Invariant Systems in the z-Domain
3.7 Summary and References
Problems
4 FREQUENCY ANALYSIS OF SIGNALS AND SYSTEMS
4.1 Frequency Analysis of Continuous-Time Signals
4.2 Frequency Analysis of Discrete-Time Signals
4.3 Properties of the Fourier Transform for Discrete-Time
Signals
4.4 Frequency-Domain Characteristics of Linear Time-Invariant
Systems
4.5 Linear Time-Invariant Systems as Frequency-Selective
Filters
4.6 Inverse Systems and Deconvolution
Problems
5 THE DISCRETE FOURIER TRANSFORM:ITS PROPERTIES AND
APPLICATIONS
5.1 Frequency Domain Sampling:The Discrete Fourier Transform
5.2 Properties of the DFT
5.3 Linear Filtering Methods Basedon the DFT
5.4 Frequency Analysis of Signals Using the DFT
5.5 Summary and References
Problems
6 EFFICIENT COMPUTATION OF THE DFT:FAST FOURIER TRANSFORM
ALGORITHMS
6.1 Efficient Computation of the DFT:FFT Algorithms
6.2 Applications of FFT Algorithms
6.3 A Linear Filtering Approach to Computation of the DFT
6.4 Quantization Effects in the Computation ofthe DFT
6.5 Summary and References
Problems
7 IMPLEMENTATION OF DISCRETE-TIME SYSTEMS
7.1 Structures for the Realization of Discrete-Time Systems
7.2 Structures for FIR Systems
7.3 Structures for IIR Systems
7.4 State-Space System Analysis and Structures
7.5 Representation of Numbers
7.6 Quantization of Filter Coefficients
7.7 Round-Off Effects in Digital Filters
7.8 Summary and References
Problems
8 DESIGN OF DIGITAL FILTERS
8.1 General Considerations
8.2 Design of FIR Filters
8.3 Design of IIR Filters From Analog Filters
8.4 Frequency Transformations
8.5 Design of Digital Filters Based on Least-Squares Method
8.6 Summary and References
Problems
9 SAMPLING AND RECONSTRUCTION OF SIGNALS
9.1 Sampling of Bandpass Signals
9.2 Analog-to-Digital ConverSion
9.3 Digital-to-Analog Conversion
9.4 Summary and References
Problems
10 MULTIRATE DIGITAL SIGNAL PROCESSING
10.1 Introduction
10.2 Decimation by a Factor D
10.3 Interpolation by a Factor I
10.4 Sampling Rate Conversion by a Rational Factor I/D
10.5 Filter Design and Implementation for Sampling-Rate
Conversion
10.6 Multistage Implementation of Sampling-Rate Conversion
10.7 Sampling-Rate Conversion of Bandpass Signals
10.8 Sampling-Rate Conversion by an Arbitrary Factor
10.9 Applications of Multirate Signal Processing
10.10 Summary and References
Problems
11 LINEAR PREDICTION AND OPTIMUM LINEAR FILTERS
11.1 Innovations Representation of a Stationary Random
Process
11.2 Forward and Backward Linear Prediction
11.3 Solution of the Normal Equations
11.4 Properties of the Linear Prediction-Error Filters
11.5 AR Lattice and ARMA Lattice-Ladder Filters
11.6 Wiener Filters for Filtering and Prediction
11.7 Summary and References
Problems
12 POWER SPECTRUM ESTIMATION
12.1 Estimation of Spectra from Finite-Duration Observations of
Signals
12.2 Nonparametric Methods for Power Spectrum Estimation
12.3 Parametric Methods for Power Spectrum Estimation
12.4 Minimum Variance Spectral Estimation
12.5 Eigenanalysis Algorithms for Spectrum Estimation
12.6 Summary and References
Problems
A RANDOM SIGNALS,CORRELATION FUNCTIONS,AND POWER SPECTRA
B RANDOM NUMBER GENERATORS
C TABLES OF TRANSITION COEFFICIENTS FOR THE DESIGN OF
LINEAR-PHASEFIRFILTERS
D LIST OF MATLAB FUNCTIONS
REFERENCES AND BIBLIOGRAPHYR1
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