Digital Signal Processing (Paperback)
S Salivahanan , A Vallavaraj , C Gnanapriy
- 出版商: McGraw-Hill Education
- 出版日期: 2001-03-09
- 售價: $900
- 貴賓價: 9.8 折 $882
- 語言: 英文
- 頁數: 805
- ISBN: 0071189823
- ISBN-13: 9780071189828
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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.
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