Signals, Systems, and Transforms by Charles L Phillips (2014-12-24)
Why Read This Book
You will get a compact, application-oriented tour of transforms and signal-processing algorithms that link theory to real engineering problems in audio, speech, radar and communications. The book emphasizes practical use of Fourier, Laplace and z-transforms, FFTs, wavelets, adaptive filtering and statistical methods so you can move quickly from analysis to implementation and testing.
Who Will Benefit
Practicing engineers and advanced undergraduates who know basic signals and systems and want to apply DSP algorithms to audio/speech, radar or communications projects.
Level: Intermediate — Prerequisites: Single-variable calculus, basic linear algebra, elementary probability and random processes, introductory signals & systems (LTI systems, impulse response), and basic programming experience (MATLAB/Python/C).
Key Takeaways
- Apply Fourier, Laplace and z-transforms to analyze continuous- and discrete-time LTI systems
- Design and implement FIR and IIR digital filters for audio, communications and radar tasks
- Compute efficient FFTs and perform practical spectral analysis and windowing for real signals
- Implement wavelet and time–frequency transforms to analyze nonstationary signals
- Develop and deploy adaptive filtering algorithms (LMS, RLS) for noise cancellation and system identification
- Use statistical signal-processing tools for detection, estimation and receiver design in communications and radar
Topics Covered
- 1. Introduction: Signals, Systems, and Engineering Perspective
- 2. Mathematical Tools: Complex Numbers, Linear Algebra and Probability Refresher
- 3. Continuous-Time Signals and LTI Systems
- 4. Laplace Transform and Continuous-Time System Analysis
- 5. Fourier Series and Fourier Transform for Continuous Signals
- 6. Sampling, Reconstruction and the Discrete-Time Domain
- 7. Discrete-Time Fourier Transform and z-Transform
- 8. FFT Algorithms and Efficient Spectral Computation
- 9. Digital Filter Design: FIR and IIR Methods, Windowing and Implementation
- 10. Spectral Analysis and Parametric Estimation
- 11. Wavelets and Time–Frequency Methods
- 12. Adaptive Filtering: LMS, RLS and Applications
- 13. Statistical Signal Processing: Detection and Estimation
- 14. Applications: Audio & Speech Processing, Radar Signal Processing, Communications
- 15. Appendices: Tables, MATLAB/Python Examples and Implementation Notes
Languages, Platforms & Tools
How It Compares
Covers much of the same practical ground as Proakis & Manolakis' DSP texts but is more transform- and application-focused; compared with Oppenheim & Willsky it trades some theoretical depth for hands-on algorithms and examples in audio, radar and communications.












