Recent speech coding technologies and standards

Abstract

This chapter presents an overview of recent developments in conversational speech coding technologies, important new algorithmic advances, and recent standardization activities in ITU-T, 3GPP, 3GPP2, MPEG and IETF that offer a significantly improved user experience during voice calls on existing and future communication systems. User experience is determined by speech quality, hence network operators are very concerned about quality of speech coders. Operators are also concerned about capacity, hence coding efficiency is another important measure. Advanced speech coding technologies provide the capability to both improve coding efficiency and user experience. One option to improve quality is to extend the audio bandwidth from traditional narrowband to wideband (16 kHz sampling) and super-wideband (32 kHz sampling). Another method is in increasing the robustness of the coder against transmission errors. Error concealment algorithms are used which substitute the missing parts of the audio signal as far as possible. In packet-switched applications (VoIP systems), special mechanisms are included in jitter buffer management (JBM) algorithms to maximize sound quality. It is of high importance to ensure standardization and deployment of speech coders that meet quality expectations. As an example of this, we refer to the Enhanced Voice Services (EVS) project in 3GPP that is developing the next generation speech coder in 3GPP. The basic motivation for 3GPP to start the EVS project was to extend the path of codec evolution by providing superwideband experience at around 13 kb/s and better quality for music and mixed content in conversational applications. Optimized behavior in VoIP applications is achieved through the introduction of high error robustness, jitter buffermanagement, inclusion of source-controlled variable bit rate operation, support of various audio bandwidths, and stereo.