- When a desired sugnal is encompassed by a noisy environment, active noise. The presented algorithm is based on the standard Least mean Squares (LMS) algorithm developed by Bernard Widrow. Modifications to the LMS algorithm were made in order ot optimixe its performance in extracting a desired speech signal form a noisy environment. The system consists of two adaptive systems running in paralles. withone having a much higher convergence rate to provide rapid adaptation in a non-stationary environment. However, theoutput of the higher converging system results indistorted speech. Therefore, the second system, which runs at a lower convergence rate but regularly has its coeffients updated by the first system, provides the actual output of the desired signal. All of the algorithm development and simulation were initially performed in matlab, and were then implemented on TMS320C6416 Digital Signal Processor(DSP) evaluation board to produce a real-time, noise-reduced speech signal.
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- Design and Implementation of an Adaptive LMS- based Parallel System for Noise Cancellation
Kevin S. Biswas
Jason G. Tong
- Springer Netherlands