Abstract
Abstract
In telephony system, the received signal by the loudspeaker, is reverberated through the environment and picked up by the microphone. It is called an echo signal. Which is in the form of time delayed and attenuated image of original speech signal, and causes a reduction in the quality of the communication. Adaptive filters are a class of filters that iteratively alter their parameters in order to minimize a difference between a desired output and their output. In the case of acoustic echo, the optimal output is an echoed signal that accurately emulates the unwanted echo signal. This is then used to negate the echo in the return signal. The better the adaptive filter simulates this echo, the more successful the cancellation will be. This paper examines LMS algorithm of adaptive filtering and the application in acoustic echo cancellation system. Employing a discrete signal processing in Matlab for simulation with real acoustic signals. Also a hardware implementation of an adaptive filter have been developed using XC3S500E Xilinx FPGA chip, and VHDL language on RTL abstraction level.
Keywords: Acoustic echo cancellation, Adaptive Filter, FPGA, VHDL.
