Voice In Real Time Compression And Decompression And Dsp Implementation

In communication systems, the digital signal possesses a much better quality than the analog signal with its higher reliability, higher security, and lower price. However, the digital signal has its own disavantages. For example, it needs higher bandwidths than the analog signal. Consequently, in order to reduce the bandwith, we must take actions to compress the original digital signal. Scientists have been studing the speech compressing technology and have got many valuable achievements since the 1980's. Among these achievements, the G.723. 1 algorithm, which intergrates lots of advantages of other low bit rate speech coding algorithms, can help to get high voice synthesis quality at the rate of 5.3Kbit/s and 6.3Kbit/s. However, the computation of the G.723.1 algorithm is very complex and the single chip cannot provide us with many resources due to the limited buget for every chip, so we have many difficulties to put the G.723. I algorithm into pratical use. Fortunately, the DSP technology has been developed for many years and the DSP chip can achieve high speed computation and powerfld system interface, so it is feasible to realize lots of speech compressing algorithms. In this paper, we use the Texas Instrument Corporation's TMS320C541 chip to realize the G.723.1 algorithms. Chapter 1 presents the transmission and reception system of the speech signal and then introduces the generation model of the digital speech signal. Chapter 2 presents the classification of the speech compressing technology, introduces the main factors which influence the speech compressing technology greatly and presents the lately development of the speech compressing technology. Chapter 3 presents the main theory of the G.723.1 algorithm, the main coding theory which Is used by the G.723.1 algorithm and the main computation of the G.723.1 algorithm. Chapter 4 presents the implementation of the G.723.1 algorithm. Chapter S gives the summary of this paper.