Introduction to Digital Signal Processing and Filter Design was developed and fine–tuned from the author′s twenty–five years of experience teaching classes in digital signal processing. Following a step–by–step approach, students and professionals quickly master the fundamental concepts and applications of discrete–time signals and systems as well as the synthesis of these systems to meet specifications in the time and frequency domains. Striking the right balance between mathematical derivations and theory, the book features:
- Discrete–time signals and systems
- Linear difference equations
- Solutions by recursive algorithms
- Time and frequency domain analysis
- Discrete Fourier series
- Design of FIR and IIR filters
- Practical methods for hardware implementation
A unique feature of this book is a complete chapter on the use of a MATLAB® tool, known as the FDA (Filter Design and Analysis) tool, to investigate the effect of finite word length and different formats of quantization, different realization structures, and different methods for filter design. This chapter contains material of practical importance that is not found in many books used in academic courses. It introduces students in digital signal processing to what they need to know to design digital systems using DSP chips currently available from industry.
With its unique, classroom–tested approach, Introduction to Digital Signal Processing and Filter Design is the ideal text for students in electrical and electronic engineering, computer science, and applied mathematics, and an accessible introduction or refresher for engineers and scientists in the field.
1.2 Application of DSP.
1.3 Discrete–Time Signals.
1.4 History of Filter Design.
1.5 Analog and Digital Signal Processing.
2. Time–Domain Analysis and z Transform.
2.1 A Linear, Time–Invariant System.
2.2 z Transform Theory.
2.3 Using z Transform to Solve Difference Equations.
2.4 Solving Difference Equations Using the Classical Method.
2.5 z Transform Method Revisited.
2.6 Convolution Revisited.
2.7 A Model from Other Models.
2.9 Solution Using MATLAB Functions.
3. Frequency–Domain Analysis.
3.2 Theory of Sampling.
3.3 DTFT and IDTFT.
3.4 DTFT of Unit Step Sequence.
3.5 Use of MATLAB to Compute DTFT.
3.6 DTFS and DFT.
3.7 Fast Fourier Transform.
3.8 Use of MATLAB to Compute DFT and IDFT.
4. Infinite Impulse Response Filters.
4.2 Magnitude Approximation of Analog Filters.
4.3 Analog Frequency Transformations.
4.4 Digital Filters.
4.5 Impulse–Invariant Transformation.
4.6 Bilinear Transformation.
4.7 Digital Spectral Transformation.
4.8 Allpass Filters.
4.9 IIR Filter Design Using MATLAB.
4.10 Yule–Walker Approximation.
5. Finite Impulse Response Filters.
5.2 Linear Phase Fir Filters.
5.3 Fourier Series Method Modified by Windows.
5.4 Design of Windowed FIR Filter Using MATLAB.
5.5 Equiripple Linear Phase FIR Filters.
5.6 Design of Equiripple FIR Filters Using MATLAB.
5.7 Frequency Sampling Method.
6. Filter Realizations.
6.2 FIR Filter Realizations.
6.3 IIR Filter Realizations.
6.4 Allpass Filters in Parallel.
6.5 Realization of FIR and IIR Filters Using MATLAB.
7. Quantized Filter Analysis.
7.2 Filter Design–Analysis Tool.
7.3 Quantized Filter Analysis.
7.4 Binary Numbers and Arithmetic.
7.5 Quantization Analysis of IIR Filters.
7.6 Quantization Analyis of FIR Filters.
8. Hardware Design Using DSP Chips.
8.2 Simulink and Real–Time Workshop.
8.3 Design Preliminaries.
8.4 Code Generation.
8.5 Code Composer Studio.
8.6 Simulator and Emulator.
9. MATLAB Primer.
9.2 Signal Processing Toolbox.