DSP Audio Compression System Based On G.723.1

This paper discussed the ITU-T G.723.1 dual rate speech coder algorithm analysis and optimizations and the real-time implementation for G.723.1 codec. For services like VoIP and video telephony,the multimedia data on the channel have to be compacted because the very limited bandwidth.Because the G.723.1 codec has a lower bit-rate and good reconstruction speech quality,it is suitable for multimedia communications in speech coding transmission. In this paper,the real-time G.723.1 audio compression system use TI High-Performance TMS320DM642 digital multimedia chip as the core implementation of a DSP-based embedded real-time voice compression and transmission system.First,this paper describe the main technology of speech coding. Such as linear predictive coding, by synthesis analysis, feeling weighted filter, multi-pulse excitation linear prediction coding, code excited linear prediction coding are discussed. Because this real-time system is based on DSP,this paper also give a brief introduction to TI C6x series chips.Second, this paper implement an real-time speech compress and transmission system which has been tested by subjective test and objective test.the objective test can pass through the ITU-T test vectors which can prove that my porting and optimizations for G.723.1 codec is successful.At last,It is proved that the real-time audio compression and transmission system is achieved by judging the real-time playback audio.