Abstract
The desire for instantaneous communication at any time and place has been a long standing dream for people of different races and cultures. The post-war progress of telecommunications technology has made such dreams a reality. Nowadays, most households in the developed world are fitted with telephone sets capable of communicating with anyone, across the globe. Over the last decade or so the demand for these communication services has seen a sharp rise. One factor that is beginning to constain these desires is the available natural resources. Limited bandwidth and high public demand have resulted in a change from primitive analog based systems to new sophisticated digital systems. The ability to transmit information at varying bit rates (hence varying capacity) has been a major step forward in conquering the problems of channel capacity. In the case of signals such as speech, high bandwidth reductions typically result in quality degradations. Such side effects can be resolved with the use of powerful Digital Signal Processing chips, allowing complex modelling of the speech signals. In any low bit rate digital speech encoding system a mathematical modelling of the signal is required. The complexity of the model is reflected in the algorithms output quality, delay, robustness to errors and computational load. The earliest versions of digital encoding techniques, such as Pulse Code Modulation, are very simple and effective. However, they operate at high bit rates of 64 kb/s, thus occupying large channel capacities. Lately, the need for efficient speech encoding has resulted in complex time domain coding schemes known as Analysis-by-Synthesis algorithms. Such complex schemes are very successful in meeting the quality objectives at low bit rates of around 6 kb/s and above. In this thesis we look at several Analysis-by-Synthesis schemes and examine their problems in meeting certain criteria such as Quality, Complexity, Robustness and Delay. The algorithms examined are all time domain techniques with their main applications in Mobile environments and PSTN services. The quality issue is assessed by looking at three major Analysis-by-Synthesis techniques, (MPE-LPC, RPE-LPC and CELP, introduced in the early eighties), which utilise different glottal excitation modelling techniques. The question of robustness in mobile applications is tackled by discussion of appropriate Forward Error Correcting codes and frame substitution/reconstruction strategies. A current requirement of PSTN services is for low delay algorithms to avoid echo effects and additional delay impairments caused via use of satellite links. Since low bit rate digital schemes incorporate linear prediction techniques resulting in long buffering and thus longer delays, backward prediction modelling is examined for achieving low delay, toll quality coding at bit rates of 8 kb/s and above.