Digital Signal Processing - Adaptive Filter Design Method Simulation Source Code

The simulation source code for adaptive filter design methods presented in Xidian University's graduate-level English textbook "Digital Signal Processing (Part 2)" is now available. This codebase was independently developed and thoroughly validated for correctness. The implementation includes key adaptive filtering algorithms such as LMS (Least Mean Squares) and RLS (Recursive Least Squares) with proper parameter tuning and convergence mechanisms. These MATLAB-based simulations demonstrate practical implementation of adaptive filter design techniques, featuring modular code structure that allows for easy modification of filter parameters, step sizes, and adaptation criteria. The code includes comprehensive error handling and performance evaluation modules to analyze filter convergence behavior and steady-state performance. Beyond digital signal processing applications, this validated codebase can be adapted for various engineering domains including communication systems, audio processing, biomedical signal analysis, and real-time system implementations. The source code provides valuable learning resources and practical implementation examples for researchers and engineers working with adaptive filtering techniques.