Table of contents for Analog and digital signal processing : an integrated, computational approach with MATLAB / John Kronenburger, John Sebeson.

Bibliographic record and links to related information available from the Library of Congress catalog.

Note: Contents data are machine generated based on pre-publication provided by the publisher. Contents may have variations from the printed book or be incomplete or contain other coding.

Table of Contents
Preface and Acknowledgments 
1.	An Introduction to Signal Processing
1.1.	Some Signal Processing History
1.2.	The Signal Processing System
2.	Describing Signals 
2.1.	Representation of Signals 
2.2.	Classification of Signals 
2.3.	Mathematical Description of Specific Signals 
3.	Continuous-Time and Discrete-Time Systems 
3.1.	General Properties of Signal Processing Systems 
3.2.	The Time-Domain Description of Linear Time-Invariant Systems
3.3.	Determination of LTI System Output by Convolution
3.4.	Transform Analysis of Continuous-Time Systems: The Laplace Transform
3.5.	Transform Analysis of Discrete-Time Systems: The Z-transform
4.	The Frequency Domain of Signals and Systems
4.1.	The Fourier Transform
4.2.	The Discrete-Time Fourier Transform 
4.3.	Calculations with the Discrete-Time Fourier Transform 
4.4.	Effects of Signal Length and Windowing on the Discrete-Time Fourier Transform 
4.5.	The Discrete Fourier Transform 
4.6.	Inverse Transforms 
4.7.	Signal Power in the Time and Frequency Domains 
4.8.	Random Noise in Signals 
4.9.	The Frequency Response of a Linear Time-Invariant DSP System
5.	Finite Impulse Response Filter Design
5.1.	General Concepts of FIR Filter Design 
5.2.	Phase Distortion and Linear Phase
5.3.	The Ideal Window Design Method 
5.4.	Sampling Design of FIR Filters
5.5.	Optimal FIR Design Methods in MATLAB
6.	Infinite Impulse Response Filter Design
6.1.	General Concepts of IIR Filter Design 
6.2.	Design by Pole-Zero Location 
6.3.	Digital Realization of Classical Analog Filters 
6.4.	MATLAB IIR Design Tools 
6.5.	Coefficient Quantization with IIR Filters 
7.	Over-Sampling and Multi-Rate DSP Systems
7.1.	Digital Anti-Aliasing 
7.2.	Down-sampling and Decimation 
7.3.	Up-Sampling and Interpolation 
7.4.	Sampling Rate Conversion by Rational Factors
7.5.	Over-Sampling and Noise
7.6.	Delta-Sigma Quantization
8.	Correlation and Auto-correlation of Signals
8.1.	The Cross-Correlation of Signals
8.2.	Auto-correlation
8.3.	Using Auto-correlation to Detect Signals in Noise 
8.4.	Detecting and Ranging a Return Echo Contaminated with Noise
9.	Adaptive Filters 
9.1.	Theory of Adaptive Filters 
9.2.	The Adaptive Predictor
9.3.	Adaptive System Identification
10.	Basic Digital Signal Processing of Images
10.1.	The Structure of Digital Images 
10.2.	Image Sampling, Quantization, and Aliasing
10.3.	Arithmetic Operations on Image Matrices
10.4.	Statistical Properties and Enhancement of Images
10.5.	Image Filtering 
10.6.	Discrete Fourier Transform of Images
11.	Wavelets 
11.1.	Non-Stationary Signals
11.2.	Sub-Band Decomposition and Reconstruction of Signals 
11.3.	Analysis of Signals Using Wavelets 
11.4.	Signal Compression Using Wavelets
12.	Case Studies in Digital Signal Processing
12.1.	Dual-Tone Multi-Frequency Signaling 
12.2.	A DSP Impedance Bridge
12.3.	JPEG Compression of Images 
12.4.	Wavelet Compression of Fingerprint Images
A.	Complex Numbers 
A.1.	Imaginary Numbers 
A.2.	Why We Need Imaginary Numbers 
A.3.	Complex Numbers 
A.4.	Polar Form of a Complex Number and Euler¿s Equation 
A.5.	Magnitude and Angle of a Complex Number 
A.6.	Complex Conjugate 
A.7.	Complex Exponential Forms of the Sine and Cosine Functions 
A.8.	Complex Functions 
A.9.	Working With Complex Numbers 
B.	A-to-D and D-to-A Conversion Methods
C.	What Makes a DSP a DSP?

Library of Congress Subject Headings for this publication:

Signal processing -- Digital techniques -- Textbooks.
Linear integrated circuits.
Signal processing -- Digital techniques -- Mathematics.