臺灣博碩士論文加值系統

在目前盛行的網際網路與個人通訊系統中，所傳遞的資料包含了影像、聲音、電子郵件等等，如何在有限頻寬的通道中傳遞大量的資料，這就需要利用資料壓縮的技術。在語音資料壓縮技術中，選擇一低傳輸率又具合理音質的編碼器是重要考量之一。碼本激發線性預估(CELP)編碼器便是符合此兩項條件的語音壓縮技術，它的架構包含了線性預估模型、音高週期搜尋與隨機碼本搜尋三個主要模組，線性預估模型是用來模擬人類聲道發聲的變化，音高週期的搜尋是用以找出類似語音訊號的週期性，而隨機碼本搜尋則是找出代表人類語音信號中雜亂的非週期性信號。本篇論文先用Matlab程式模擬標準CELP演算法，使其位元率在6.9 kbps，經過主觀的聆聽和客觀的數據判斷下得到合理的語音品質，並試著找出減低演算法複雜度的方法，然後利用C語言實現演算法於德州儀器公司所發展的C6201定點數數位信號處理器模擬板(EVM)上，選用定點數數位信號理器雖然有低耗電量、低成本的優點，但在發展軟體的過程卻必須花更多的注意力在溢位問題及大小的調整。而即時實現編解碼的技巧是在中央處理單元不干涉的情況下利用兩個直接記憶體存取(DMA)通道控制資料的接收及傳送，並使用雙緩衝區作乒乓效應(ping-pong effect)的處理，經由實際量測編碼速度得知，數位信號處理器在DMA存取完ping緩衝區之前，就已處理完pong緩衝區，反之亦然。如此一來兩個緩衝區可輪流接收語音資料，符合即時的要求。

Recently, in popular internet and personal communication system, it transmits information including image, voice and E-mail data etc.. How can we transmit these large information in channels of limited bandwidth? It needs to use of techniques of data compression. Therefore, in the speech data compression aspects, it is important to choose the coder with low transmission bit rate and reasonable speech quality. CELP coder just matches these two conditions. There are three components in the CELP architecture. One is linear predictive module, it is used to simulate the variants of human vocal tract. Two is the procedure of searching pitch lag, it is used to find the quasi-periodicity of speech signal. Three is procedure of searching random codebook, it is used to find the close white noise of speech signal. In this thesis, we simulate the standard CELP algorithm with Matlab program and let bit rates in 6.9 kbps. We get resonable speech quality according to listen subjectively and measure SNR objectively. Then we try to find the method of reducing complexity of algorithm. Aterward, we implement the algorithm to TI C6201 DSP evaluation module board with C language. Indeed it has advantage of low power consumption and cost. But we must take lots of time to concentrate the problem of overflow or size scaling. For encoding in real time, we use two DMA channels to control data receive from and transmit to without CPU intervention. Then we use dual buffers as processing of ping-pong effect. By measuring the speed of encoding and decoding, it appears that DSP has already finished pong buffer processing before DMA finished ping buffer saving. It is the same otherwise. So two buffers can receive speech data in turn and fullfill requirements in real time.

Chap.1 緒論................................................1
1.1 研究背景............................................1
1.2 研究動機............................................4
1.3 研究方法............................................5
1.4 章節概要............................................5
Chap.2 數位語音編碼之原理與技術............................6
2.1 前言................................................6
2.2 語音模型的建立......................................6
2.3 線性預估編碼(Linear Predictive Coding)..............7
2.3.1 LPC參數的求解...................................9
2.3.2 LPC參數的量化...................................11
2.4 音高週期的預測(Pitch Prediction)....................12
2.5 碼激式線性預估(CELP)編碼器介紹......................15
2.5.1 經由合成分析法(Analysis-by-Synthesis)...........15
2.5.2 標準CELP演算法..................................17
2.5.3 碼本搜尋的方法..................................18
Chap.3 Matlab 模擬標準CELP演算法...........................20
3.1 前言................................................20
3.2 CELP演算法架構......................................20
3.2.1 LPC的分析.......................................22
3.2.2 音高週期的預測與加速運算的方法..................27
3.2.3 隨機碼本的設計..................................28
3.2.4 位元配置........................................29
3.2.5 解碼端架構......................................30
3.3 客觀性音質評估......................................30
3.3.1 聽覺信號雜訊比之量測評估........................31
3.3.2 實驗結果........................................32
Chap.4 TMS320 C6201 EVM板 介紹.............................36
4.1 前言................................................36
4.2 TMS320C6201 處理器架構..............................36
4.2.1 中央處理單元....................................37
4.2.2 記憶體配置......................................38
4.2.3 內部周邊裝置....................................40
4.3 C6201模擬板之硬體架構...............................46
4.3.1 初始設定........................................46
4.3.2 外部記憶體......................................47
4.3.3 聲音輸出入介面..................................49
4.4 軟體發展環境........................................49
Chap.5 用EVM板實現即時CELP語音編解碼.......................52
5.1 前言................................................52
5.2 CELP演算法定點化分析................................52
5.2.1 資料格式定位處理................................52
5.2.2 線性預估分析....................................54
5.2.3 全極點濾波器分析................................56
5.3 程式架構與最佳化的方法..............................56
5.3.1 利用DMA模式抓取資料.............................56
5.3.2 雙緩衝器互換處理................................57
5.3.3 建立固定表格代替計算............................57
5.3.4 利用C6201兩組運算單元...........................60
5.4 效能量測............................................61
Chap.6 結論與未來展望......................................64
參考文獻...................................................65

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