Digital Signal Processing (Paperback)

S Salivahanan , A Vallavaraj , C Gnanapriy




This book comprehensively covers the undergraduate course on Digital Signal Processing. Computer usage is integrated into the text in the form of problem solving using MATLAB. Solved examples and critical-thinking exercises and review questions enhance the reader's comprehension of the concepts.


Table of Contents:

1.  Classification of Signals and Systems

     1.1 Introduction

     1.2 Classification of Signals

     1.3 Singularity Functions

     1.4 Amplitude and Phase Spectra

     1.5 Classification of Systems

     1.6 Simple Manipulation of Discrete-time Signals

     1.7 Fepresentations of Systems

     1.8 Analog-to-Digital Conversion of Signals

     Review Questions

2.  Fourier Analysis of Periodic and Aperiodic Continuous-Time Signals and Systems

     2.1 Introduction

     2.2 Trigonometric Fourier Series

     2.3 Complex or Exponential form of Fourier Series

     2.4 Parseval's Identity for Fourier Series

     2.5 Power Spectrum of a Periodic Function

     2.6 Fourier Transform

     2.7 Properties of Fourier Transform

     2.8 Fourier Transform of Some Important Signals

     2.9 Fourier Transform of Power and Energy Signals

     Review Questions

3.  Applications of Laplace Transform to System Analysis

     3.1 Introduction

     3.2 Definition

     3.3 Region of Convergence (ROC)

     3.4 Laplace Transforms of Some Important Functions

     3.5 Initial and Final Value Theorems

     3.6 Convolution Integral

     3.7 Table of Laplace Transforms

     3.8 Partial Fraction Expansions

     3.9 Network Tansfer Function

     3.10 s-plane Poles and Zeros

     3.11 Laplace Transform of Periodic Functions

     3.12 Application of Laplace Transformation in Analysing Networks

     Review Questions

4.  z-Transforms

     4.1 Introduction

     4.2 Definition of the Z-transform

     4.3 Properties of z-transform

     4.4 Evaluation of the Inverse z-transform

     Review Questions

5.  Linear Time Invariant Systems

     5.1 Introduction

     5.2 Properties of a DSP Systems

5.3 Difference Equation and its Relationship with System Function, Impulse Response and Frequency Response

     5.4 Frequency Response

     Review Questions

6.  Discrete and Fast Fourier Transforms

     6.1 Introduction

     6.2 Discrete Convolution

     6.3 Discrete-Time Fourier Transform (DTFT)

     6.4 Fast Fourier Transform (FFT)

     6.5 Computing an Inverse DFT by Doing a Direct DFT

     6.6 Composite-radix FFT

     6.7 Fast (Sectioned) Convolution

     6.8 Correlation

     Review Questions

7.  Finite Impulse Response (FIR) Filters

     7.1 Introduction

     7.2 Magnitude Response and Phase Response of Digital Filters

     7.3 Frequency Response of Linear Phase FIR Filters

     7.4 Design Techniques for FIR Filters

     7.5 Design of Optimal Linear Phase FIR Filters

     Review Questions

8.  Infinite Impulse Response (IIR) Filters

     8.1 Introduction

     8.2 IIR Filter Design by Approximaton of Derivatives

     8.3 IIR Filter Design by Impulse Invariant Method

     8.4 IIR Filter Design by the Bilinear Transformation

     8.5 Butterworth Filters

     8.6 Chebyshev Filters

     8.7 Inverse Chebyshev Filters

     8.8 Eilliptic Filters

     8.9 Frequency Transformation

     Review Questions

9.  Realisation of Digital Linear Systems

     9.1 Introduction

     9.2 Basic Realisation Block Diagram and the Signal-flow Graph

     9.3 Basic Structures for IIR Systems

     9.4 Basic Structures for FIR Systems

     Review Questions

10. Effects of Finite World Length in Digital Filters

     10.1 Introdctuon

     10.2 Rounding and Truncation Errors

    10.3 Quantisation Effects in Analog-to-Digital Conversion of Signals

     10.4 Output Noise Power from a Digital System

    10.5 Coefficient Quantisation Effects in Direct Form Realisation of IIR filters

    10.6 Coefficient Quantisation in Direct Form Realisation of FIR Filters

     10.7 Limit Cycle Oscillations

     10.8 Product Quantisation

     10.9 Scaling

     10.10 Quantisation Errors in the Computation of DFT

      Review Questions

11. Multirate Digital Signal Processing

     11.1 Introduction

     11.2 Sampling

     11.3 Sampling Rate Conversion

     11.4 Signal Flow Graphs

     11.5 Filter Strucrtures

     11.6 Polyphase Decomposition

     11.7 Digital Filter Design

     11.8 Multistage Decimators and Interpolators

     11.9 Digital Filter Banks

     11.10 Two-channel Quadratrue Mirror Filter Bank

     11.11 Multilevel Filter Banks

     Review Questions

12. Spectral Estimation

     12.1 Introduction

     12.2 Energy Density Spectrum

    12.3 Estimation of the Autocorrelation and Power Spectrum of Random Signals

     12.4 DFT in Spectral Estimation

     12.5 Power Spectrum Estimation: Non-Parametric Mehtods

     12.6 Power Spectrum Estimation: Parametric methods

      Review Questions

13. Adaptive Filters

     13.1 Introduction

     13.2 Examples of Adaptive filtering

     13.3 The Minimum Mean Square Error Criterion

     13.4 The Widrow LMS Algorithm

     13.5 Recursive Least Square Algorithm

     13.6 The Forward-Backward Lattice Method

     13.7 Gradient Adaptive Lattice Method

     Review Questions

14. Applications of Digital Signal Processing

     14.1 Introduction

     14.2 Vocie Processing

     14.3 Applications to Radar

     14.4 Applications to Image Processing

     14.5 Introduction to Wavelets

     Review Questions

15. MATLAB Programs

     15.1 Introduction

     15.2 Representation of Basic Signals

     15.3 Discrete Convolution

     15.4 Discrete Correlation

     15.5 Stability Test

     15.6 Sampling Theorem

     15.7 Fast Fourier Transform

     15.8 Butterworth Analog Filters

     15.9 Chebyshev Type-1 Analog Filters

     15.10 Chebyshev Type-2 Analog Filters

     15.11 Butterworth Digital IIR Filters

     15.12 Chebyshev Type-1 Digital Filters

     15.13 Chebyshev Type-2 Digital Filters

     15.14 FIR Filter Design Using Window Techniques

     15.15 Upsampling of a Sinusoidal Signal

     15.16 Down Sampling a Sinusoidal Sequence

     15.17 Decimator

     15.18 Estimation of Power Spectral Density (PSD)

     15.19 PSD Estimator

     15.20 Periodogram Estimation

     15.21 State-space Representation

     15.22 Partial Fraction Decomposition

     15.23 Inverse z-transform

     15.24 Group Delay

     15.25 Overlap-add Method

     15.26 IIR Filter Design-impulse Invariant Mehtod

     15.27 IIR Filter Design-bilinear Transformation

     15.28 Direct Realisation of IIR Digital Filters

     15.29 Parallel Realisation of IIR Digital Filters

     15.30 Cascade Realisation of Digital IIR Filters

     15.31 Decimation by Polyphase Decomposition

     15.32 Multiband FIR Filter Design

     15.33 Analysis Filter Bank

     15.34 Synthesis Filter Bank

     15.35 Levinson-Durbin Algorithm

     15.36 Wiener Equation's Solution

     15.37 Short-time Spectral Analysis

15.38 Cancellation of Echo produced on the Telephone Base Band Channel

15.39 Cancellation of Echo Produced on the Telephone Pass Band Channel

     Review Questions






1.  信號和系統的分類

     1.1 簡介

     1.2 信號的分類

     1.3 奇異函數

     1.4 振幅和相位頻譜

     1.5 系統的分類

     1.6 離散時間信號的簡單操作

     1.7 系統的表示

     1.8 信號的模擬轉換


2.  周期性和非周期性連續時間信號和系統的傅立葉分析

     2.1 簡介

     2.2 三角傅立葉級數

     2.3 傅立葉級數的複數或指數形式

     2.4 傅立葉級數的Parseval恆等式

     2.5 周期函數的功率頻譜

     2.6 傅立葉變換

     2.7 傅立葉變換的性質

     2.8 一些重要信號的傅立葉變換

     2.9 功率和能量信號的傅立葉變換<```