A new source coding technique called MODULO-PCM (MPCM) is presented and it shown that this new scheme has essentially the same performance as linear predictive coding or transform coding. In contrast with the conventional schemes, MPCM employs a simple memoryless encoder and a moderately complex decoder incorporating the Viterbi algorithm. Bounds for distortion in MPCM systems fora first-order Gauss-Markov process are numerically calculated.

Adaptive Differential Pulse Code Modulation (ADPCM) systems can provide high‐quality digitizations of telephone‐bandwidth speech at a bit rate of 32 kb/s. At a lower bit rate such as 24 kb/s, the quality of the speech is limited by an easily perceptible level of quantization noise. This paper proposes an adaptive postfiltering procedure that can significantly enhance the quality of lower bit rate ADPCM. The coefficients of the postfilter are easily derivable from the predictor coefficients in the ADPCM decoder. In a subjective test involving 18 listeners and two sentence‐length test inputs, the enhanced 24‐kb/s speech with an optimized postfilter design ranks very close to conventional 32‐kb/s speech. A suggested application of the postfiltering procedure is in packet voice or mobile radio systems where substandard bit rates such as 24 kb/s or 16 kb/s are sometimes necessary. The postfiltering algorithm has also been successfully tested in non‐DPCM situations, such as in the enhancement of speech degraded by additive white Gaussian noise.

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