A split signal polynomial as a model of an impulse noise filter for speech signal recovery

The synthesis of the non-linear non-recursive digital filter of impulse noise on the basis of the splitting method in time domain is described. The filter recovers speech signals, distorted by impulse noise. The filter model is constructed as the splitting polynomial of an odd degree. The splitter is the time delay line, comprising the equal number of previous and subsequent samples with respect to the current time moment. The polynomial parameters result from solving an approximation problem in the mean-square norm. It is shown that the filter with the splitting model provides more precise speech signal recovery than the median and Volterra filters.