Study and Design of VoIP using G.729 Algorithm

The algorithm described below is the 8 kb/s speech coding algorithm. The algorithm is based on a CS-ACELP (Conjugate-Structure Algebraic Code Excited Linear Prediction) coding technique. Short term synthesis filter is based on a 10th order Linear Prediction filter for every 10ms frame. This coder is designed to operate with a digital signal obtained by first performing telephone bandwidth filtering of the analogue input signal, then sampling it at 8000 Hz, followed by conversion to 16-bit linear PCM for the input to the encoder. The output of the decoder should be converted back to an analogue signal by similar means. The idea of the algorithm is to predict the next coming signals by means of linear prediction. It also uses statistical data to distinguish the resemblance of the signal to special signals in its codebook. It uses the fixed codebook and adaptive codebook. Long-term pitch synthesis filter is implemented using the adaptive-code book. The codec delivers quality of speech equivalent to that of 32 kb/s ADPCM (Adaptive Delta Pulse Code Modulation) for most operating conditions. This algorithm gives the reduced bit rates at the cost of lower quality. The coder structure will be discussed in detail and also the reasons behind certain design choices.