High-Gain Digital Speech Interpolation with Adaptive Differential PCM Encoding

Abstract

This paper describes a digital speech interpolationadaptive differential PCM bit reduction technique in which digital speech interpolation (DSI) is combined with ADPCM encoding. A highly sensitive speech detector, a voiceband data discriminator, and a variable rate ADPCM encoding are used to achieve a high compression ratio. The speech detector proposed in [1] detects speech signals above -51 dBm with 32 ms hangover time; average talk spurt activity of 36 percent was measured on fully loaded trunks in an international satellite link. Features of the speech power spectrum are used for adaptively controlling the bit length from 2 to 4 in an ADPCM speech encoder. Voiceband data are detected with 10 ms by the voiceband data discriminator. 5 bit ADPCM encoding is applied to voiceband data to maintain transparency through the DSI-ADPCM system. A DSI gain of 3 is expected as a result of the highly sensitive speech detection, the variable rate encoding technique, and the voiceband data discrimination. Speech and voiceband data are efficiently transmitted through an ADPCM encoding with either a 6 or 6.4 kHz sampling rate converted from an 8 kHz sampling rate. To avoid a band limitation as much as possible, a frequency shift manipulation on the voiceband channel is incorporated prior to the sampling conversion. Consequently, a total bit reduction gain of 7 to 4 is expected relative to a 64 kbit/s PCM transmission. Satisfactorily high quality of the processed speech has been obtained through computer simulations.