Audio Signals Noise Cancellation using Adaptive LMS algorithm

Project Description

Signal processing is an operation designed for extracting, enhancing, storing, and transmitting useful information. Hence signal processing tends to be application dependent. In contrast to the conventional filter design techniques, adaptive filters do not have constant filter coefficients and no priori information is known. Such a filter with adjustable parameters is called an adaptive filter. Adaptive filter adjust their coefficients to minimize an error signal and can be realized as finite impulse response (FIR), infinite impulse response (IIR), lattice and transform domain filter. The most common form of adaptive filter is the transversal filter using least mean square (LMS) algorithm In this project LMS algorithm is implemented in which step followed for implementation are as • Firstly have audio signal from user • Signal will be mixed with noise • Noisy signal is given to adaptive filter (LMS) • Filtration is performed by adaptive filter • Final de-noised signal is obtained • Analysis is done by comparing de-noised signal with original signal