# Digital Signal Processing (Paperback)

### S Salivahanan , A Vallavaraj , C Gnanapriy

• 出版商:
• 出版日期: 2001-03-09
• 售價: \$900
• 貴賓價: 9.8\$882
• 語言: 英文
• 頁數: 805
• ISBN: 0071189823
• ISBN-13: 9780071189828
• 立即出貨 (庫存=1)

## 商品描述

Description:

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.

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

6.7 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

13.1 Introduction

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

Review Questions

14. Applications of Digital Signal Processing

14.1 Introduction

14.2 Vocie Processing

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

15.26 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