Digital Signal Processing Using 1st edition by Vinay Ingle, John Proakis ISBN 1337026123 9781337026123

Ch 1: Introduction

1.1: Overview of Digital Signal Processing

1.2: A Brief Introduction to MATLAB

1.3: Applications of Digital Signal Processing

1.4: Brief Overview of the Book

Ch 2: Discrete-Time Signals and Systems

2.1: Discrete-Time Signals

2.2: Discrete Systems

2.3: Convolution

2.4: Difference Equations

2.5: Problems

Ch 3: The Discrete-Time Fourier Analysis

3.1: The Discrete-Time Fourier Transform (DTFT)

3.2: The Properties of the DTFT

3.3: The Frequency Domain Representation of LTI Systems

3.4: Sampling and Reconstruction of Analog Signals

3.5: Problems

Ch 4: The z-Transform

4.1: The Bilateral z-Transform

4.2: Important Properties of the z-Transform

4.3: Inversion of the z-Transform

4.4: System Representation in the z-Domain

4.5: Solutions of the Difference Equations

4.6: Problems

Ch 5: The Discrete Fourier Transform

5.1: The Discrete Fourier Series

5.2: Sampling and Reconstruction in the z-Domain

5.3: The Discrete Fourier Transform

5.4: Properties of the Discrete Fourier Transform

5.5: Linear Convolution Using the DFT

5.6: The Fast Fourier Transform

5.7: Problems

Ch 6: Implementation of Discrete-Time Filters

6.1: Basic Elements

6.2: IIR Filter Structures

6.3: FIR Filter Structures

6.4: Overview of Finite-Precision Numerical Effects

6.5: Representation of Numbers

6.6: The Process of Quantization and Error Characterizations

6.7: Quantization of Filter Coefficients

6.8: Problems

Ch 7: FIR Filter Design

7.1: Preliminaries

7.2: Properties of Linear-Phase FIR Filters

7.3: Window Design Technique

7.4: Frequency-Sampling Design Technique

7.5: Optimal Equiripple Design Technique

7.6: Problems

Ch 8: IIR Filter Design

8.1: Some Preliminaries

8.2: Some Special Filter Types

8.3: Characteristics of Prototype Analog Filters

8.4: Analog-to-Digital Filter Transformations

8.5: Lowpass Filter Design Using MATLAB

8.6: Frequency-Band Transformations

8.7: Problems

Ch 9: Sampling Rate Conversion

9.1: Introduction

9.2: Decimation by a Factor D

9.3: Interpolation by a Factor I

9.4: Sampling Rate Conversion by a Rational Factor I/D

9.5: FIR Filter Designs for Sampling Rate Conversion

9.6: FIR Filter Structures for Sampling Rate Conversion

9.7: Problems

Ch 10: Round-Off Effects in Digital Filters

10.1: Analysis of A/D Quantization Noise

10.2: Round-Off Effects in IIR Digital Filters

10.3: Round-Off Effects in FIR Digital Filters

10.4: Problems

Ch 11: Applications in Adaptive Filtering

11.1: LMS Algorithm for Coefficient Adjustment

11.2: System Identification or System Modeling

11.3: Suppression of Narrowband Interference in a Wideband Signal

11.4: Adaptive Line Enhancement

11.5: Adaptive Channel Equalization

Ch 12: Applications in Communications

12.1: Pulse-Code Modulation

12.2: Differential PCM (DPCM)

12.3: Adaptive PCM and DPCM (ADPCM)

12.4: Delta Modulation (DM)

12.5: Linear Predictive Coding (LPC) of Speech

12.6: Dual-Tone Multifrequency (DTMF) Signals

12.7: Binary Digital Communications

12.8: Spread-Spectrum Communications

Ch 13: Random Processes

13.1: Random Variable

13.2: A Pair of Random Variables

13.3: Random Signals

13.4: Power Spectral Density

13.5: Stationary Random Processes through LTI Systems

13.6: Useful Random Processes

13.7: Summary and References

Ch 14: Linear Prediction and Optimum Linear Filters

14.1: Innovations Representation of a Stationary Random Process

14.2: Forward and Backward Linear Prediction

14.3: Solution of the Normal Equations

14.4: Properties of the Linear Prediction-Error Filters

14.5: AR Lattice and ARMA Lattice-Ladder Filters

14.6: Wiener Filters for Filtering and Prediction

14.7: Summary and References

Ch 15: Adaptive Filters

15.1: Applications of Adaptive Filters

15.2: Adaptive Direct-Form FIR Filters

15.3: Summary and References

Bibliography

Index