A Digital System for Multiplex Speech Transmission

Abstract

This thesis describes the investigation of a digital transmission system intended for multichannel telephony. The object of the system is to achieve a bendwidth economy compared with conventional PCM (Pulse Code Modulation) systems, while maintaining good voice quality. This is accomplished by using simple predictive techniques to remove redundant samples from the digitized speech signal. A computer aided study of conversational speech revealed that such a strategy would eliminate e substantial fraction of samples. The transmitted digit-rate is reduced, compared with PCM, by having fewer timeslots in the frame of transmitted pulses than channels. Mutual interference between channels occurs when the number of samples to be transmitted exceeds the number of available timeslots. This is termed frame overflow and places an upper limit on the number of channels for a given number of timeslots. Some methods of minimising the effects of frame overflow are described. The performance of these methods are compared by deriving signal-to-overflow noise ratios. It is concluded that the number of channels can be doubled, compared with conventional PCM, for the same bandwidth. A simulation model of the proposed transmission system was constructed. Signal-to-noise ratio measurements and subjective tests on the model confirmed the findings of the theoretical analysis.