Voice Signals Analysis Tool

Abstract

Last few years had witnessed an explosive growth in the field of digital communication. One of the principle techniques enabling this growth is voice/speech coding in which an analog signal from a microphone is digitally sampled via an A-to–D converter and then efficiently compressed into a digital bit stream for transmission or storage. A corresponding voice decoder receives these bit stream samples, which are suitable for playback through Dto- A converter and a loudspeaker. Speech coders are of different forms each of which differ in terms of bit rate (degree of compression), complexity (MIPS and Memory) and voice quality. There are a large number of voice coding techniques which are being used presently e.g. RELP, CELP, STC, MBE, ADM, ADPCM, LPC, VQ etc. The theme of thesis is to implement ADPCM voice coding technique and develop and implement software to identify the different versions of ADPCM technique like G.711, G.721, and G.723 at various compression rates. These codes were studied, analyzed, implemented and appropriate code was made for the respective identifications and conversions. Matlab was also used for their verifications.