| In 21 century, communication should supply all kinds of good ways to satisfypeople's need in exchanging information conveniently. Because most people transferinformation by speech, which is the most important, efficient and convenient way incommunication, speech communication holds the importance in modern digitalcommunication system. However the sources in communication especially thefrequency source become more and more valuable with the development ofinformation society. Then more and more speech compress coding technologies comeinto being to decrease bit rate in transmission and data storage quantity when inmemory.High speed DSPs(Digital signal processors)make it possible to realize thereal real-time speech coding system. After many years rapidly development of speechcompression technology, it becomes more and more important.This paper is to introduce an algorithm of low bit rate speech coding –MBE.MBE is a parameter coding method based on analysis-synthesis. It subdivides speechframe to several sub-bands and estimates parameters in sub-bands. Then based on thevoice/unvoice judgement of every sub-bands to decide apply different excited sourcein composing speech. So it can get more precise parameters of speech to improve thequality and robustness of speech in decoding part and the complexity of the algorithmis not increase apparently. We amended the pitch smooth algorithm in pitch firstestimation phase, improved the estimating precision of speech pitch and the judgmentvoiced and unvoiced speech. At the same time reduce the delay because of the speechcoding.At last, paper will present the design of low bit rate speech coding real-timesystem based on TMS320C6416DSP of TI Company.The system include CCS IDEsoftware and other DSP's hardware peripherals including: AD /DA module-TLV320AIC23, CPLD, SDRAM and Flash.This low bit rate speech coding system isbased on radio digital data transformitting station and can make digital datatransformiting station support speech transmission. |
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