A new adaptive method for the coding of audio signals is proposed in this paper, where the audio signal is sampled in the amplitude domain and not in the time domain, as is done in the traditional methods. The resulting encoded output is a bit stream of variable rate, following the bandwidth variations of the source. The proposed system consists of a coder and a decoder. Both parts apply a differential amplitude sampler employing a predictor to remove the redundancy from the signal. The predictor used is adaptive, optimizing its operation at regular intervals by observing the statistics of the signal. Both parts of the model are described and results are presented that show the performance of the system in comparison to previous analogous methods.
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