語音編碼不論是在資料壓縮或是語音的傳輸上都極為重要，而語音傳輸是目前最主要也最普遍的通訊傳輸服務。儘管語音編碼技術很早就發展出來，但是隨著科技的發展依舊不斷的推陳出新。新的語音編碼傳輸位元率已由8Kbps（CELP）與4.8Kbps（CS-ACELP）發展至2.4Kbps（MELP）。也因為傳輸位元率的降低，語音品質就只能由更複雜的演算法來提升，這使得實現快速語音編碼相當困難。
繼FS1016 CELP 4.8K bps之後, 美國國防部於1996年宣佈了混合激發線性預測編碼器(MELP)，為2.4K bps語音編碼的新標準，其目的是在傳輸網路頻寬限制下，藉由壓縮後資料的傳送，來節省傳輸位元率，並在解碼端能迅速還原音質良好的聲音訊號。
因此本論文也特別針對MELP語音編碼器架構之各單元作一介紹。並著手於程式的撰寫，程式部分是以C語言來實現，且列舉幾個例子來比較合成語音與原始語音的差異，結果驗證合成語音與原始語音以人耳辨識幾乎一樣。最後，提出語音壓縮系統未來之研究方向。

Speech coding is very important in data compression and speech communication. Speech communication is at present the most dominant and common service in telecommunication networks. Though, speech coding technology is well developed, science and technology still to find new ways of speech coding from old theories continuously. The transmission bit rate of new digital speech coding techniques has dropped from 8 Kbps (CELP), 4.8Kpbs (CS-ACELP) to 2.4 Kbps (MELP). As the bit rate decreasing, the speech quality can only be maintained by employing very complex algorithms which are difficult to implement in real time.
After FS1016 CELP 4.8K bps, the MELP vocoder was selected as the new 2400 bps Federal Standard speech coder by the United Stated Department of Defense Digital Voice Processing Consortium (DDVPC) in 1996. The purpose is in limited bandwidth of the transmission network by data transfer after compression, to economize bit rates, and to return to high quality of the original speech signal quickly in the decoder.
Hence, the units of MELP coder will be described in this thesis. The algorithm will be programmed in C language. We list several examples to compare the difference between the synthesized speech and the original speech. Experiments show that the synthesized speech sounds almost identical to the original speech. Finally, research direction in the future of speech compression system will be discussed.

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