Design and Implementation of FPGA Based LMS Self-Adjusting Adaptive Filtering System for Audio Signal Processing

摘要

This paper presents the design and implementation of an adaptive filter using the state-of-the-art Xilinx Vivado software/hardware co-design concepts and tools. A desired signal corrupted by the environment can often be recovered by an adaptive noise canceller using the least mean squares (LMS) algorithm. The detailed structure of the adaptive noise cancellation system is illustrated. The adaptive parameters of the least-mean-square based adaptive filter system are obtained using the MATLAB/Simulink model. RTL design is generated by converting LMS design in Simulink to an Intellectual Property (IP) Core using HDL Coder Support. A complete system of Filter based on Zynq board target architecture is designed using Vivado Synthesis Design and VHDL target language. The IP Core is adopted in Vivado Synthesis and implementation. Finally, the debugger is run before the audio file was fed in Zedboard development board for test. Experimental results show that the proposed hardware implementation method has a high degree of noise cancellation performance.