I asked this question on the DSP forum and got a reasonable answer but the answer was very technical and was wondering whether someone could give me a more general idea of how it works. My understanding of it is basically:
LPC works by digitising the analog signal and splitting it into segments. For each of the segments we determine the key features of the signal and try to encode these as accurately as possible.
The key features are the pitch of the signal (i.e. the basic formant frequency), the loudness of the signal and whether the sound is voiced or unvoiced. Parameters called vocal tract excitation parameters are also determined which are used in the vocal tract model to better model the state of the vocal tract which generated the sound.
This data is passed over the network and decoded at the receiver. The pitch of the signal is used as input to either a voiced or unvoiced synthesiser, and the loudness data is used to boost the amplitude of this resulting signal. Finally the vocal cord model filters this sound by applying the LPC coefficients which were sent over the network.
What I'm finding hard to understand is the LPC coefficients - what exactly are they and what is going on with the linear predictive coding i.e. using the past samples etc. Any help is much appreciated since I have an exam on this on tuesday!