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Adaptive Signal Processing, 1e

NA 2002


Why Read This Book

You will learn the theory and practical tools to design, analyze, and implement adaptive filters for real-world problems in audio, speech, radar, and communications. The book balances rigorous convergence and statistical analysis with algorithmic insights and application examples so you can move from math to working prototypes quickly.

Who Will Benefit

Graduate students, signal-processing engineers, and R&D practitioners who already know basic DSP and want to apply adaptive filtering and statistical signal-processing techniques to audio, radar, and communications problems.

Level: Advanced — Prerequisites: Undergraduate signals & systems and linear algebra, probability and random processes (basic estimation theory), familiarity with discrete-time DSP concepts and comfort with MATLAB or equivalent numerical computing environments.

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Key Takeaways

  • Implement and tune common adaptive algorithms such as LMS, NLMS, and RLS for time-varying signal environments.
  • Analyze convergence, steady-state error, and stability of adaptive filters using mean‑square and statistical tools.
  • Design and apply adaptive schemes for audio/speech tasks (echo cancellation, noise suppression) and communications/radar problems (channel equalization, clutter suppression).
  • Perform spectral analysis and FFT-based processing and integrate wavelet methods with adaptive filtering for nonstationary signals.
  • Use adaptive beamforming and array-processing concepts to enhance spatial selectivity in radar and communication receivers.

Topics Covered

  1. 1. Introduction to Adaptive Signal Processing and Applications
  2. 2. Review of Linear Algebra, Random Processes, and Estimation Basics
  3. 3. Optimum Linear Filters and the Wiener Solution
  4. 4. Stochastic Gradient Methods: LMS and Variants
  5. 5. Recursive Least Squares and Fast RLS Algorithms
  6. 6. Performance Analysis: Convergence, Misadjustment, and Tracking
  7. 7. Frequency-Domain and FFT-Based Adaptive Filtering
  8. 8. Wavelets, Time–Frequency Methods, and Adaptive Multiresolution Techniques
  9. 9. Adaptive Array Processing and Beamforming
  10. 10. Applications: Audio/Speech Enhancement, Echo Cancellation, Communications Equalization, Radar Signal Processing
  11. 11. Kalman Filtering and State-Space Adaptive Methods
  12. 12. Implementation Issues, Real-Time Considerations, and Case Studies

Languages, Platforms & Tools

MATLABOctavePython (NumPy/SciPy)General-purpose DSP processors (conceptual coverage)PC-based prototyping and audio I/O hardwareReal-time embedded targets (conceptual)MATLAB Signal Processing Toolbox / SimulinkNumPy/SciPyFFTW (for FFT implementations)

How It Compares

Covers much of the same practical and theoretical ground as Haykin's Adaptive Filter Theory but places stronger emphasis on application examples (audio, radar, communications) and implementation trade-offs; more applied than classic texts like Widrow & Stearns.

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