The Study And Optimization Of AMR-WB Based On TM1300

The current second-generation and third-generation (3G) mobile communication systems operate with narrow audio bandwidth limited to 200-3400Hz.As wireless systems are evolving from voice-telephony dominated services to multimedia and high-speed data services, the introduction of a wider bandwidth of 50–7000 Hz speech codec which provides substantially improved speech quality and naturalness is quite necessary. The Adaptive Multi-rate Wideband (AMR-WB) speech codec which was selected by 3GPP/ETSI and ITU-T can provide natural, present and comfortable codec speech and have higher intelligence and robustness in wireless systems. A wide range of applications are envisioned for AMR-WB, including ISDN wideband telephony and audio/video teleconferencing,Voice over IP and Internet applications such as IP video conferencing, voice mail, voice chat, broadcast, and voice streaming.This paper describes AMR-WB codec and its implementation optimization based on TM1300. Firstly, The principle of AMR-WB algorithm is discussed in detail, Secondly a brief introduction of the widely used audio/video DSP–TM1300,which is presented by Philip Corp, is given.Then we concentrate on the techniques of the optimization of AMR-WB algorithms based on TM1300. Finally, the result of the experiment is given out to show that distinctive improvement is obtaineded in efficiency and performance, and it is important for the implementation of AMR-WB in commerce.