滤波器系数 Resources

Showing items tagged with "滤波器系数"

This document presents the schematic diagram of a linear adaptive equalizer (refer to "Introduction to Modern Digital Signal Processing" p.275 for illustration) implemented using the LMS algorithm. Key implementations include plotting the mean-squared error convergence curve for a single trial, deriving final filter coefficients, and conducting 20 independent experiments with 500-sample training sequences. Comparative analysis of three distinct step-size values is provided, demonstrating algorithm stability and convergence behavior through MATLAB-based simulations.

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This study considers a linear adaptive equalizer's principle block diagram, referenced from the adaptive equalizer application schematic on page 275 of "Introduction to Modern Digital Signal Processing". The implementation utilizes the LMS (Least Mean Squares) algorithm to achieve adaptive equalization, featuring convergence curves of squared errors from single experiments and final filter coefficients. Key implementation aspects include a 500-sample training sequence length, 20 independent trials for statistical analysis, and performance comparison across three distinct step-size parameters to evaluate convergence characteristics.

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Source code for computing all-pass filter coefficients based on channel group delay, designed to correct signal phase variations with algorithm implementation details

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Simulation of Farrow filters for fractional delay adjustment, implementing 4th-order, 8th-order, and 16th-order filter configurations. The M-file contains coefficient generation functions while the MDL file demonstrates post-filtering deviation estimation scenarios with parameter tuning and performance analysis.

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