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First Principles of Discrete Systems and Digital Signal Processing

Strum, Robert D., Kirk, Donald E. 1988

This textbook presents both discrete systems and digital signal processing in a conversational style that relies on a minimum of mathematics. The authors use carefully crafted pedagogy and detailed examples to improve students' problem solving skills, to help them see interrelationships and connections, and to integrate new material with what they have seen in previous chapters. The book also provides a number of computer-based methods for solving problems.


Why Read This Book

You should read this book if you want a clear, low‑math introduction to the core ideas of discrete systems and digital signal processing that emphasizes intuition and problem solving. You'll get conversational explanations, worked examples, and computer‑based approaches that make it easier to apply DSP ideas to audio, radar, and communications problems without getting bogged down in heavy proofs.

Who Will Benefit

Undergraduate or early‑graduate students and practicing engineers seeking a practical, example‑driven foundation in discrete systems and DSP to support work in audio/speech, radar, or communications.

Level: Intermediate — Prerequisites: Single‑variable calculus, basic linear algebra, and introductory circuit or signals & systems concepts (familiarity with sinusoids, complex numbers, and basic differential equations recommended).

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Key Takeaways

  • Explain the fundamental properties of discrete‑time signals and LTI discrete systems using intuitive, example‑based reasoning.
  • Apply the z‑transform, discrete‑time Fourier transform (DTFT), and convolution to analyze and design discrete systems.
  • Compute and use DFT/FFT techniques for efficient spectral analysis and implement basic spectral estimation procedures.
  • Design and compare common digital filters (FIR and IIR), including windowing methods and practical considerations for stability and implementation.
  • Use computer‑based numerical techniques and pseudocode to simulate DSP algorithms and validate signal‑processing designs.
  • Relate statistical signal processing and adaptive filtering concepts to practical problems in audio/speech and communications.

Topics Covered

  1. 1. Introduction to Discrete Signals and Systems
  2. 2. Sampling, Quantization, and Signal Representation
  3. 3. Time‑Domain Analysis and Convolution
  4. 4. The z‑Transform and System Function
  5. 5. Frequency Analysis: DTFT and Frequency Response
  6. 6. The Discrete Fourier Transform and FFT Algorithms
  7. 7. Windowing and Practical Spectral Analysis
  8. 8. Digital Filter Design: FIR Techniques
  9. 9. Digital Filter Design: IIR Techniques and Realization Forms
  10. 10. Numerical and Computer‑Based Methods for DSP
  11. 11. Statistical Signal Processing and Estimation Basics
  12. 12. Adaptive Filters and Applications
  13. 13. Applications: Audio/Speech, Radar, and Communications
  14. 14. Case Studies, Problem‑Solving Strategies, and Further Reading

Languages, Platforms & Tools

PseudocodeMATLAB/Octave (recommended for examples)C (implementation examples)General numerical computing platforms (no hardware‑specific focus)Numerical methods and simulation (suitable for MATLAB/Octave or generic scientific computing environments)FFT libraries / algorithmic descriptions (implementable in common languages)

How It Compares

Compared with Oppenheim & Schafer or Proakis & Manolakis, Strum's text is noticeably more conversational and example‑driven with less emphasis on rigorous proofs and more on practical problem solving.

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