臺灣博碩士論文加值系統

數位化科技的進步，促使過去眾多以類比訊號表現聲音方式紛紛改採數位擷取方式呈現，由於在擷取數位訊號的過程中，常常因為編碼器數位取樣化的不連續特性，造成聲音訊號品質與原始訊號上的失真，為了補償原始聲音類比訊號在轉換時的失真，常常需要將頻率濾出再加以補償或衰減。基於上述原因，本文探討無線脈衝響應雙二階濾波器(IIR Biquad Filter)特性，將所需要的頻率濾出，然後利用八位元微處理器算出雙二階濾波器所需係數數值，再將係數數值送至德州儀器數位訊號處理器(Digital Signal Processor)內部之雙二階濾波暫存器，藉以達到改善聲音頻率的補償和調整。同時本文探討和改善人耳所能感受之音頻範圍20Hz至20KHz的訊號失真。

The coming digital-era leads the voice signal, which was represented by analog format in the past, to be converted into digital format. In the process of digitalization, the discontinuous characteristic of sampling induces the distortion of voice. To overcome this distortion, several frequency bands of voice signal are needed to be filtered, and then, to be amplify or to be attenuated at the same time.
Base on the above reasons, the characteristics of IIR Biquad Filter are investigated first. And then, a filter is used to filter out wanted frequency. To implement the improvement of voice quality by using a filter, the parameters of IIR Biquad Filter are calculated by 8-bits microcontroller and sent to a digital signal processor that includes built-in IIR Biquad Filter.
This thesis focused on the distortion in the frequency band which is audible to human ears.

中文摘要 i
英文摘要 ii
誌謝 iii
表目錄 vii
圖目錄 viii
第一章 序論 1
1.1研究動機 1
1.2研究目標 2
1.3論文架構 2
第二章 濾波器設計原理與分析 3
2.1濾波器基本原理 3
2.2濾波器特性參數 5
2.3數位化濾波特性 7
2.4雙二階濾波器 10
2.4.1低通濾波器 12
2.4.2高通濾波器 14
2.4.3帶通濾波器 16
2.4.4帶拒濾波器 18
2.4.5等化濾波器 20
第三章 數位訊號處理器功能特性說明 23
3.1 系統方塊圖與區塊功能說 23
3.2 TAS3204系統架構 24
3.2.1 TAS3204主要功能 25
3.2.2 TAS33204系統與程式設計 26
3.3 AT89C51ED2系統式設計 28
3.3.1 AT89C51ED2主要功能 29
3.3.2 AT89C51ED2系統與程式設計 30
3.4 USB DAC 37
3.5 S/PDIF編解碼器應用 37
3.6 PC端軟體連線 39
3.7 通用序列匯流排集線器 40
3.8 系統電源 41
第四章 數位訊號處理器實驗結果 43
4.1數位訊號處理器實驗步驟 43
4.2數位訊號處理器模擬輸出入實驗結果 47
4.3類比訊號處理實驗結果 54
4.4數位訊號處理實驗結果 62
第五章 結論 70
5.1結論 70
5.2未來工作 70
參考文獻 71
簡歷 74

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