Sampled speech compression system

Abstract

A sampled speech compression and expansion system, for two-dimensional prssing of speech or other type of audio signal, comprises transmit/encode apparatus and receive/decode apparatus. The transmit/encode apparatus comprises a low-pass filter, adapted to receive an input signal, for passing through low-frequency analog signals. A converter is connected to the low-pass filter for converting the analog signal into a digital signal. A buffer memory, whose input is connected to the converting means, stores the digitized signals. A correlator, having inputs from the A/D converter and the buffer memory, correlates the digital signal received directly from the converter with a delayed signal from the buffer memory. An 'interval-select' circuit, whose input is connected to the output of the correlator, uses the autocorrelation value as a basis for comparison with subsequent peaks in the correlation value which are greater than a specified fraction of the autocorrelation value. The interval-select circuit has an output which is connected to the buffer memory, the value of the fractional peaks and their timing being stored in the buffer memory. A transform circuit, whose input is connected to the buffer memory, performs an even discrete cosine transform (EDCT) of the stored signal. A first modulator, whose input is connected to the output of the EDCT means, differentially pulse code modulates (DPCM) its input signal. A second modulator, whose input is connected to the output of the interval select circuit, differentially pulse code modulates its input signal. A multiplexer, having an input connected to the output of the first and second modulating means, combines the two differentially pulse code modulated signals. A receiver/decoder has circuits which perform an inverse function to those of the transmitter/coder and are arranged in inverse order, from input to output, to those of the transmitter/coder.