Design And Implement Of The Voice Codec System Based On DSP

With the rapid development of Chinese communication industry, it asks that the graduates should not only have strong knowledge of communication theory, but also have the comprehensive design and innovation capability. In order to meet the actual need of the community to foster practical and muti-level talents, the school intends to develop a modern communication experimental system for the undergraduate teaching in the experiment.Starting from the request of the modern communication experimental system, the subject have designed and implemented a voice codec system based on DSP(digital signal processor). On request of the system, this paper has completed designs of the hardware and software. The system has implemented the real-time PCM&ADPCM processing of voice signal on DSP. In addition, it has designed the PC-USB2.0-DSP architecture using the GPIF(General Programmable Interface) mode, which belongs to CY7C68013A of CYPRESS company. According to this architecture, this subject has also finished the PC application programming. What'more, it has still designed each functional module's drive programming. And ultimately, PC's real-time controlling DSP through the USB2.0 interface to do voice or data communications has been implemented. In addition, the voice and data stream can be implemented the seamless connection between the digital interface on the board and the channel part of the modern communication experimental system. After testing the voice codec system, all the functions have met the design requirements.Meanwhile, to meet the broad application requirement of the USB interface technology, the paper also has used another interface mode of CY7C68013A, that is the Slave FIFO mode, and designed a general USB2.0 interface, which can be used for any one of MCU. The design has been successfully used in the development of a chemical testing instrument. Both the hardware and software are working correctly, and there is no error during data transmission.