Lowdelay combined wideband speech and audio coding with scalable bit rates

Abstract

The integration of fixed and wireless networks have expanded the range in which speech and audio coders were designed to operate. As a result, current research is focusing on a number of developments including algorithms which are able to adapt to different transmission environments and to operate under multiple constraints of bit rate, complexity, delay, robustness to bit errors and diversity of input signals. In this paper, we propose a single coding algorithm for compressing wideband speech and audio signals (0-8kHz) operating with scalable bit rates and with low delay. The algorithm is based on the backward-adaptive linear predictive coding (BA LPC) technique in conjunction with an efficient closedloop optimised excitation structure consisting of sparse pulses of ternary values. The output bit rates range from 17 to 68kb/s. The scalability feature is achieved by means of discrete quantisation layers representing various levels of enhancements of the base-line coder and also flexibility in terms of complexity and bit allocation requirements depending on the particular application and on the network resources. An evaluation of the performance of the coder operating at 17kb/s is carried out using the G.722 standard as a reference.