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研究生: 洪彥瑋
Hung Yen-Wei
論文名稱: 語音降噪演算法之研究
The study of a new noise cancellation algorithm
指導教授: 黃奇武
Huang, Chi-Wu
學位類別: 碩士
Master
系所名稱: 工業教育學系
Department of Industrial Education
論文出版年: 2005
畢業學年度: 94
語文別: 中文
論文頁數: 123
中文關鍵詞: 數位訊號處理嵌入式系統主動式噪音控制離散餘弦轉換語音增強
英文關鍵詞: digital signal processing, embedded system, active noise control, discrete cosine transform, speech enhancement
論文種類: 學術論文
相關次數: 點閱:367下載:66
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  • 本論文的研究重點在於提出了一個基於噪音偵測、消除以及透過線性平滑濾波器抑制殘餘噪音的語音增強演算法。
    在本論文中,我們特別強調要處理的是與語音訊號頻率分佈相近的同頻域噪音消除演算法。本演算法源於主動式噪音控制和語音增強濾波器的觀念,但於噪音插入點偵測以及頻譜刪減演算法當中具有其獨創性。在論文當中首先說明如何克服訊號漂移的困難,利用統計噪音訊號穩定度的觀念先建構出一個穩定的噪音樣本。接著發展出一個透過此穩定的噪音樣本與被噪音污染的訊號源能量分析而能夠偵測噪音源插入點的『Diff-RMS』演算法,以及在搜尋到插入點之後透過離散餘弦轉換將噪音與被污染訊號皆轉換至頻域上做噪音消除的動作。而此一消噪後的訊號則經由離散餘弦逆轉換將其轉換回時域當中。最後則透過一個線性的平滑濾波器將聲音訊號中的殘餘噪音能量抑制到人耳不可聽聞之程度。全部過程當中同時具有語音訊號在時域和頻域的處理觀念,以及許多聲學上的數位訊號處理技巧。這些也是我們論文中會一一探討的部分。
    最後並將本噪音消除演算法實現於嵌入式系統發展平台上,驗證其能夠穩定的運行,並且進一步符合未來實用的價值。

    The practical thrust of this thesis is to propose a new speech enhancement algorithm based on noise detection/cancellation and depress the remnant noise by a linear smooth filter.
    In this thesis, we put special emphasis on that the frequency distribution of noise we want to cancel is the same as the frequency distribution of speech signal in our algorithm. The new algorithm based on the concept of active noise control and speech enhancement filter, but has the originality in noise corrupted point detection and spectral-subtraction process. First of all, we explain how to win through the difficulty of signal drift by using the method of construct a stable noise template with statistics. Second, we develop a noise detection algorithm named "Diff-RMS algorithm" which analyze the energy of noise and noise corrupted signal. After we find the correct noise corrupted point, we adaptively subtract the coefficients of discrete cosine transform of noise template form an input signal. When subtraction is done, we convert the entire processed coefficients form frequency domain back to time domain by inverse discrete cosine transform. In the end, we apply a linear smooth filter to depress the remnant noise to an inaudible degree.All of the thetechnological processes contains acoustics concepts of digital signal processing in both time domain and frequency domain and that will be discussed in the following chapter in the thesis.
    Finally, we implement the new noise cancellation algorithm on embedded platform to verify the stability and future practicability.

    第一章 緒論 6 第一節 研究背景 6 第二節 研究動機 8 第三節 研究目的 9 第四節 研究方法與步驟 9 第五節 論文組織與大綱 10 第二章 文獻探討 12 第一節 濾波器與雜訊消除 12 第二節 主動式噪音控制 22 第三章 研究系統架構 35 第一節 簡介 35 第二節 Creator-S3C4510系統簡介 39 第三節 語音相關之軟硬體 44 第四節 演算法系統架構 47 第四章 音訊前處理與研究理論 51 第一節 聲音聽覺理論 51 第二節 聲音訊號之特性 56 第三節 音頻訊號資料之前處理 60 第四節 離散餘弦轉換 73 第五節 語音品質評量 77 第五章 演算法系統架構 83 第一節 消噪演算法系統架構 83 第二節 高頻噪音消除演算法 84 第三節 同頻域之定樣噪音消除演算法 86 第四節 嵌入式平台實現 100 第五節 實驗結果 106 第六章 結論與未來展望 113 第一節 研究結論 113 第二節 未來展望 114 參考文獻 116

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