Book description
Updated and expanded, the second edition of this bestselling text introduces the fundamentals of DSP. Along with discussing current DSP applications, this edition includes new chapters on analog systems models and pattern recognition using support vector machines as well as new sections on the chirp z-transform, resampling, waveform reconstruction, discrete sine transform, and logarithmic and nonuniform sampling. It also contains a more comprehensive table of transforms. Numerous exercises and examples harness the power of MATLAB and MATLAB functions and examples are available for download online.
Table of contents
- Cover
- Half Title
- Title Page
- Copyright Page
- Dedication
- Table of Contents
- Foreword to the Second Edition
- Foreword to the First Edition
- Preface to the Second Edition
- Preface to the First Edition
- Authors
- 1. Introduction
- 2. Least Squares, Orthogonality, and the Fourier Series
-
3. Correlation, Fourier Spectra, and the Sampling Theorem
- 3.1 Introduction
- 3.2 Correlation
- 3.3 The Discrete Fourier Transform (DFT)
- 3.4 Redundancy in the DFT
- 3.5 The FFT Algorithm
- 3.6 Amplitude and Phase Spectra
- 3.7 The Inverse DFT
- 3.8 Properties of the DFT
- 3.9 Continuous Transforms, Linear Systems, and Convolution
- 3.10 The Sampling Theorem
- 3.11 Waveform Reconstruction and Aliasing
- 3.12 Resampling
- 3.13 Nonuniform and Log-Spaced Sampling
- Exercises
- References
- Further Reading
-
4. Linear Systems and Transfer Functions
- 4.1 Continuous and Discrete Linear Systems
- 4.2 Properties of Discrete Linear Systems
- 4.3 Discrete Convolution
- 4.4 The z-Transform and Linear Transfer Functions
- 4.5 The Complex z-Plane and the Chirp z-Transform
- 4.6 Poles and Zeros
- 4.7 Transient Response and Stability
- 4.8 System Response via the Inverse z-Transform
- 4.9 Cascade, Parallel, and Feedback Structures
- 4.10 Direct Algorithms
- 4.11 State-Space Algorithms
- 4.12 Lattice Algorithms and Structures
- 4.13 FFT Algorithms
- 4.14 Discrete Linear Systems and Digital Filters
- 4.15 Functions Used in This Chapter
- Exercises
- References
- Further Reading
-
5. FIR Filter Design
- 5.1 Introduction
- 5.2 An Ideal Lowpass Filter
- 5.3 The Realizable Version
- 5.4 Improving an FIR Filter with Window Functions
- 5.5 Highpass, Bandpass, and Bandstop Filters
- 5.6 A Complete FIR Filtering Example
- 5.7 Other Types of FIR Filters
- 5.8 Digital Differentiation
- 5.9 A Hilbert Transformer
- Exercises
- References
- Further Reading
-
6. IIR Filter Design
- 6.1 Introduction
- 6.2 Linear Phase
- 6.3 Butterworth Filters
- 6.4 Chebyshev Filters
- 6.5 Frequency Translations
- 6.6 The Bilinear Transformation
- 6.7 IIR Digital Filters
- 6.8 Digital Resonators and the Spectrogram
- 6.9 The All-Pass Filter
- 6.10 Digital Integration and Averaging
- Exercises
- References
- Further Reading
-
7. Random Signals and Spectral Estimation
- 7.1 Introduction
- 7.2 Amplitude Distributions
- 7.3 Uniform, Gaussian, and Other Distributions
- 7.4 Power and Power Density Spectra
- 7.5 Properties of the Power Spectrum
- 7.6 Power Spectral Estimation
- 7.7 Data Windows in Spectral Estimation
- 7.8 The Cross-Power Spectrum
- 7.9 Algorithms
- Exercises
- References
- Further Reading
-
8. Least-Squares System Design
- 8.1 Introduction
- 8.2 Applications of Least-Squares Design
- 8.3 System Design via the Mean-Squared Error
- 8.4 A Design Example
- 8.5 Least-Squares Design with Finite Signal Vectors
- 8.6 Correlation and Covariance Computation
- 8.7 Channel Equalization
- 8.8 System Identification
- 8.9 Interference Canceling
- 8.10 Linear Prediction and Recovery
- 8.11 Effects of Independent Broadband Noise
- Exercises
- References
- Further Reading
-
9. Adaptive Signal Processing
- 9.1 Introduction
- 9.2 The Mean-Squared Error Performance Surface
- 9.3 Searching the Performance Surface
- 9.4 Steepest Descent and the LMS Algorithm
- 9.5 LMS Examples
- 9.6 Direct Descent and the RLS Algorithm
- 9.7 Measures of Adaptive System Performance
- 9.8 Other Adaptive Structures and Algorithms
- Exercises
- References
- Further Reading
-
10. Signal Information, Coding, and Compression
- 10.1 Introduction
- 10.2 Measuring Information
- 10.3 Two Ways to Compress Signals
- 10.4 Adaptive Predictive Coding
- 10.5 Entropy Coding
- 10.6 Transform Coding and the Discrete Cosine Transform
- 10.7 The Discrete Sine Transform
- 10.8 Multirate Signal Decomposition and Subband Coding
- 10.9 Time–Frequency Analysis and Wavelet Transforms
- Exercises
- References
-
11. Models of Analog Systems
- 11.1 Introduction
- 11.2 Impulse-Invariant Approximation
- 11.3 Final Value Theorem
- 11.4 Pole–Zero Comparisons
- 11.5 Approaches to Modeling
- 11.6 Input-Invariant Models
- 11.7 Other Linear Models
- 11.8 Comparison of Linear Models
- 11.9 Models of Multiple and Nonlinear Systems
- 11.10 Concluding Remarks
- Exercises
- References
- Further Reading
-
12. Pattern Recognition with Support Vector Machines
- 12.1 Introduction
- 12.2 Pattern Recognition Principles
- 12.3 Learning
- 12.4 Support Vector Machines
- 12.5 Multi-Class Classification
- 12.6 MATLAB® Examples
- Exercises
- References
- Appendix: Table of Laplace and z Transforms
Product information
- Title: Digital Signal Processing with Examples in MATLAB, 2nd Edition
- Author(s):
- Release date: April 2016
- Publisher(s): CRC Press
- ISBN: 9781000755633
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