Coding techniques based on adaptive linear prediction and quantization are well suited to signals carried by telephone channels and can provide, with a per channel rate of 32 kbits/s, a level of quality compatible with the specifications of the conventional 64 kbit/s rate. The ADPCM technique described in this paper features a simple adaptive quantization scheme and a tenth-order linear prediction adaptive filter realized as a cascade of five second-order sections. Besides superior performance characteristics, the nonaccumulation of degradations in tandem connections is achieved. The implementation is a 60 channel PCM-ADPCM converter, called TMN 162. It is a fully digital equipment which offers a doubling of the capacity of digital links in telephone networks.

A 32 kbit/s rate per channel would double the capacity of digital links compared with 62 kbit/s standard PCM and permit digital techniques to be used throughout in telecommunication networks. The requirements imposed on a coding scheme for long-haul telephone transmission are reviewed. Besides the minimal complexity of implementation, the toll-quality level, the capability of handling signals other than speech, a crucial feature is the non cumulation of degradations in successive coding-decoding processes. An ADPCM coder is proposed ; in the implementation with finite precision arithmetic, both adapted quantization step sizes and prediction filter outputs track in coder and decoder. Experimental results show that the 32 kbit/s scheme outperforms 56 kbit/s PCM for speech in several aspects and is adequate for data transmission up to 4800 bit/s.

Techniques applied to the digital processing of speech and signaling in a 60 channel FDM to 30 channel PCM converter are presented. Great efficiency is achieved through connecting a particular type of Fourier Transform computer to a polyphase network in which coefficient symmetries are exploited. Digital processing parameter values have been chosen so as to meet specified performance objectives. Realization of a laboratory model is outlined and measured performance is reported. Adequacy of the various options retained for the digital processing is confirmed by the experimental results.