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.
