Minimize interlaced noise using the Adaptive Filter

AbstractSuggested this project a filter for speech signal enhancement. While steps can be listed below : First recording speech signal from microphone. Second input signal should be divided and attached into overlapped frames. Third these frames should be identified and recognized at (Voiced Sound)frame signals and(Unvoiced Sound). By using two adaptive filter algorithms: The Least Mean Squares(LMS) algorithm and Recursive Least Squares (RLS) algorithm. Then the objective tests have been done by measuring the LMS errors between input and output signals. When the suggested filter applied to several of applied samples, the results showed the filter order has agreed influence upon filtration time, and on the square of the error signal. The tests revealed that the LMS algorithm gives good results of reducing the square of the error signal. : (Voiced Sound) Unvoiced Sound)). (Adaptive Filter): LMS RLS . . LMS RLS .