Adpcm Speech Coder with Adaptive Transmission and Arma Modelling of Non-uniformly Sampled Signals
نویسندگان
چکیده
In a previous paper [1], a coding principle based upon non-uniform transmission of signal samples was introduced. This coding principle relies on the use of an efficient prediction/reconstruction algorithm for non uniformly sampled signals in both the transmitter and the receiver. Recently, we proposed an adaptive ARMA estimation method for time series with missing samples [2]. The purpose of this paper is to study the feasibility of an ADPCM coding scheme using adaptive transmission in order to reduce the average bit rate, that could allow to share multiple lines carrying voice or to multiplex voice and data. The ITU-T G.726 ADPCM standard [3] is considered in order to evaluate the performances of the proposed speech coder.
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