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研究生：	陳豪成 Chen, Hao-Cheng
論文名稱：	以DSP為基礎之MP3解碼器的實現 Implementation of DSP-Based Decoder for MP3
指導教授：	何明字 Ho, Ming-Tzu
學位類別：	碩士 Master
系所名稱：	工學院 - 工程科學系 Department of Engineering Science
論文出版年：	2003
畢業學年度：	91
語文別：	中文
論文頁數：	76
中文關鍵詞：	解碼器、MP3 、定點式數位訊號處理器
外文關鍵詞：	decoder, fixed-point DSP, MP3
相關次數：	點閱：116 下載：2
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MPEG-1/Audio標準規格對高品質的數位音樂提供了三個層級的編解碼方式，其中又以第三層(MPEG-1/Audio Layer 3)的架構最複雜，但是在相同位元率下，以第三層的聲音品質和壓縮比為最佳，其中壓縮比可達到1：10 ~ 1：12。
本論文主要針對MPEG-1/Audio Layer 3音訊標準來作研究，並於定點運算的數位訊號處理器晶片上實現一解碼器。實現的重點包含程式的撰寫、定點數的運算、高效率的運算法則，及於硬體平台上實現出一個可播放出音樂的解碼器。在程式的部分，我們以C語言搭配組合語言來撰寫，而硬體方面使用由德州儀器所開發的TMS320VC5402 DSK來實現，此DSK的核心是一顆定點式TMS320VC5402 DSP晶片。
實現結果可發現以C語言撰寫的程式會使解碼器整體解碼效率變差，所以吾人針對解碼過程中運算量大及效率差的模組重新以組合語言撰寫，以改善執行效率。而為了達到即時的目的，吾人也針對這些運算大的模組提出一個改善效率的架構。實現結果證實可於數位訊號處理器上成功實現MPEG-1/Audio Layer 3音訊解碼器。

The MPEG-1/Audio standard offers three levels for the compression of high quality digital audio. One of them is the MPEG-1/Audio Layer 3. It is the most complex scheme but provides best sound quality and compression ratio. The compression ratio can reach 1:10 ~ 1:12.
In this thesis, a survey is first presented for the MPEG-1/Audio Layer 3 audio standard. The MP3 decoder is implemented on a fixed-point DSP chip. The key features of this decoder implementation include coding, fixed-point operation, and efficient algorithms. In coding, the algorithms of the decoder will be programmed in C and assembly. For the hardware, the MP3 decoder is developed on the TMS320C5402 DSK, which used a TMS320VC5402 DSP chip.
Experiments show coding with C language gives bad efficiency of decoder’s performance. Therefore, those inefficient modules of the decoding process are rewritten in assembly. In order to achieve real-time processing, efficient algorithms are also proposed for those inefficient modules. Finally, it is shown that a MPEG-1/Audio decoder can be successfully implemented on a DSP chip.

中文摘要	................................I
英文摘要	................................II
誌謝	................................III
目錄	................................IV
表目錄	................................VII
圖目錄	................................VIII
第一章 緒論	
  1-1 前言 ..............................1-1
  1-2 數位訊號處理器 ....................1-2
  1-3 研究動機 ..........................1-3
  1-4 相關文獻回顧	 ......................1-3
  1-5 本文結構 ..........................1-3
第二章 MPEG-1/Audio Layer 3編/解碼原理	
  2-1 前言 ..............................2-1
  2-2 MPEG-1/Audio Layer 3編碼原理 ......2-1
  2-3 多重相位分解濾波器器 ..............2-2
  2-4 離散餘弦轉換.......................2-3
  2-5 重疊現象處理 ......................2-6
  2-6 重新排列 ..........................2-7
  2-7 神經模組 ..........................2-8
  2-8 非均勻量化 ........................2-10
  2-9 霍夫曼編碼 ........................2-11
  2-10 附屬資訊編碼 .....................2-12
  2-11 位元流的格式處理 .................2-12
  2-12 MPEG-1/Audio Layer 3解碼原理 .....2-13
  2-13 同步 .............................2-14
  2-14 霍夫曼資訊解碼 ...................2-19
  2-15 霍夫曼解碼 .......................2-21
  2-16 解比例因子 .......................2-23
  2-17 反量化 ...........................2-25
  2-18 立體聲處理過程 ...................2-25
  2-19 重新排列 .........................2-27
  2-20 重疊現象處理 .....................2-27
  2-21 反離散餘弦轉換 ...................2-28
  2-22 多重相位合成濾波器 ...............2-29
第三章 硬體及軟體環境介紹	
  3-1 前言 ..............................3-1
  3-2 硬體環境 ..........................3-1
  3-3 軟體環境 ..........................3-5
第四章 架構實現與結果	
  4-1 前言 ..............................4-1
  4-2 程式架構實現流程 ..................4-1
  4-3 程式最佳化流程 ....................4-2
  4-4 定點運算的實現 ....................4-5
  4-5 實驗結果 ..........................4-9
第五章  一個有效率的架構實現與結果	
  5-1 前言 ..............................5-1
  5-2 反離散餘弦轉換的改良架構 ..........5-1
  5-3 多重相位合成濾波器模組之一有效架構
      的改良 ............................5-6
  5-4 一個有效率架構的實現結果 ..........5-8
第六章 結論與未來展望
  6-1 結論 ..............................6-1
  6-2 未來展望 ..........................6-1
參考文獻

                                    

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校內：2004-08-28公開
校外：2004-08-28公開

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