基本信息

書名:數字信號處理：原理、算法與應用(第3版 影印版)——國外經典計算機科學教材

定價：89.00元

售價：60.5元,便宜28.5元,摺扣67

作者:(美)伯卡斯,(美)拉卡斯

齣版社：中國電力齣版社

齣版日期：2004-09-01

ISBN：9787508324999

字數：

頁碼：

版次：1

裝幀：平裝

開本：

商品重量：0.4kg

編輯推薦

本書是作者在總結針對本科生和研究生階段的數字信號處理教學經驗的基礎上形成的。本書針對電子工程、計算機工程和計算機科學專業的學生介紹瞭離散時間信號、係統和現代信號處理算法的基本原理及其應用，重點集中於數字信號處理係統的分析和設計以及計算機實現。全書內容的編排既側重於理論又深入實際應用，且每章後都有許多精心設計的習題來幫助讀者掌握所學知識。

內容提要

為瞭給讀者在理論和實踐應用之間進行閤理的平衡，本書嚴謹地介紹瞭現代數字信號處理的基本概念和技術，並介紹瞭相關的算法和應用。本書涵蓋瞭綫性離散時間係統分析的時域和頻域方法，還涉及瞭諸如采樣、數字濾波器設計、濾波器實現、去捲積、插值、狀態矢量空間方法、頻譜分析等相關主題的內容。本書不僅要求對諸多示例、練習的理解，而且更強調對數字信號算法進行軟件實現的實踐環節。
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，並對其進行瞭更加閤理清晰的重組——介紹DFT，並在闡明傅立葉分析後描述其快速計算
·描述模擬信號模數轉換中涉及的運算和技術
·在時域研究綫性時不變離散時間係統和離散時間信號的特性
·考慮雙邊z變換和單邊z變換，並描述瞭求z反變換的方法
·在頻域分析信號與係統，給齣連續時間信號與離散時間信號的傅立葉級數與傅立葉變換
·實現無限衝激響應（IIR）與有限衝激響應（FIR）係統的結構形式，包括直接型、級聯型、並聯型、格型和格梯型
·采樣頻率轉換基礎與多采樣率轉換係統
·功率譜估計的詳細測試，並討論瞭非參數方法、基於模型的方法和基於特徵分解的方法，包括MUSIC算法和ESPRIT算法
·全書囊括瞭許多實例，並提供大約500個可解決的問題
本書既適閤作為本科生學習離散係統和數字信號處理課程的教材，又適閤研究生一年級學習數字信號處理課程時作為教材使用。

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 PUTATION 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
INDEX

作者介紹

John G.Proakis長期擔任美國東北大學的電氣工程教授，並擔任該校電氣與計算機工程係主任之職達14年之久。他分彆從麻省理工學院和哈佛大學獲得瞭碩士和博士學位。Proakis教授是眾多成功教材的作者，其教材在世界上具有相當的影響力。

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