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研究生: 賴旭謙
Lai, Hsu-Chien
論文名稱: 應用聲源定位與感知計算來決策以相干性為基準之雙麥克風噪音抑制演算法與其實現
Novel Coherence-Based Noise Reduction Algorithm By Applying Sound-Source-Localization And Awareness-Computation Strategy For Dual Microphone
指導教授: 雷曉方
Lei, Sheau-Fang
共同指導教授: 賴信志
Lai, Shin-Chi
學位類別: 碩士
Master
系所名稱: 電機資訊學院 - 電機工程學系
Department of Electrical Engineering
論文出版年: 2014
畢業學年度: 102
語文別: 中文
論文頁數: 86
中文關鍵詞: 噪音抑制(NR)聲源定位(SSL)感知計算(AC)快速傅立葉轉換(FFT)
外文關鍵詞: Noise Reduction(NR), Sound Source Localization(SSL), Awareness-Computation(AC), Fast-Fourier Transform(FFT)
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  • 本篇論文提出了利用聲源定位系統整合以相干性為基準之噪音抑制演算法,用以改善現有文獻無法抑制前方噪音的缺點以及在後方噪音的環境下會有估計上的誤差。聲源定位系統是利用雙麥克風間的交互相關性(Cross-Correlation),透過計算交互相關性最大值來求得訊號的時間延遲差(Time Difference of Arrival , TDOA),進而能判斷噪音源方向,同時,我們加入適應性(adaptive)的臨界值估測,用以降低誤判機率。此做法(Proposed)不僅可以處理環境中複雜多變的噪音源,而且以相干性為基準的噪音抑制也具有簡單實現的優點。在整合上,我們提出以感知計算(Awareness-Computation , AC)與演算法共模組化來降低運算複雜度,使得乘法運算複雜度僅提升9.09%。不管在穩定噪音或非穩定噪音的環境下,噪音抑制演算法都不需要任何噪音估計或語音動態偵測器(Voice Activity Detector , VAD),可大量減少計算上造成的複雜度。礙於麥克風距離的物理限制,傳統以相干性為基準之噪音抑制演算法無法適用且實作於助聽器上,而本論文採用之噪音抑制演算法係依雙麥克風之間的SNR推導所得,因此能有效地應用於助聽器上。根據軟體模擬後的實驗結果得知,我們提出的噪音抑制效果能比現有文獻在SNR的評比上高出3dB。綜合上述,我們的演算法具備了運算複雜度低、能有效抑制穩定噪音與非穩定噪音以及不用任何的噪音估計等三種優點並且可符合數位助聽器的硬體限制,因此更適合應用於數位助聽器及搭載於智慧型手機等小型裝置上。

    This paper presents a novel coherence-function-based noise suppression algorithm (NSA) with a weighted overlap-add (WOLA) filterbank for dual microphones. It solved the following two issues: One is that traditional method cannot efficiently suppress the noise from the front, and the other is that it may cause estimation errors while the noise is from the back. Consider the complexity in algorithm; the proposed method employs a simple sound source localization (SSL) algorithm, and an awareness computation (AC) strategy to improve these drawbacks instead of using complex voice active detector (VAD). By calculating the cross-correlation of dual microphones, the information of time difference of arrival (TDOA) is obtained. Hence, the direction of noise source can be effectively estimated. To reduce the error rate of finding out the exact noise source, an adaptive threshold value is introduced. From the view of system integration, the AC and module-sharing scheme are also adopted to reduce the computational complexity. The results show that the number of multiplication of the proposed method is only 9.09% increased, and the SNR of the proposed algorithm has at least 3dB growth which is higher than that of other approaches. In FPGA implementation, the proposed SSL design can be operated at 25 MHz which is easily to achieve the real-time requirement of 72.625 kHz. Overall, it is very suitable for integrating with Fourier-transform-based WOLA hearing aid design in the future.

    中文摘要 I EXTENDED ABSTRACT III 誌謝 XI 目錄 XIII 表目錄 XV 圖目錄 XVII 第一章 緒論 1 1.1. 噪音抑制之背景簡介 1 1.1.1. 基本介紹 1 1.1.2. 噪音種類的介紹 4 1.1.3. 噪音抑制的演算法介紹 6 1.1.3.1. 頻譜相減法(Spectral Subtractive Algorithms)[6] 6 1.1.3.2. 波束形成器(Beamformer Algorithms)[7-11] 8 1.1.3.3. 相干性之噪音抑制演算法(Coherence-Based Algorithm)[12-19] 12 1.2. 研究目的與研究動機 15 1.3. 論文章節組織 17 第二章 相關文獻的介紹與分析 19 2.1. Yousefian et al. Coherence-Based演算法[18] 19 2.1.1. 相干性函數之基本定義與推導 19 2.1.2. 噪音抑制演算法之增益函數設計與分析 22 2.1.3. 文獻探討 25 2.2. Yousefian et al. SNR-Estimator演算法[19] 27 2.2.1. 演算法介紹 27 2.2.2. SNR估計之推導公式 28 2.2.3. 文獻探討 31 2.3. 結論 35 第三章 利用聲源定位與感知計算整合以相干性為基準之噪音抑制演算法 37 3.1. 以交互相關性為基準之聲源定位演算法 37 3.1.1. 基本概念介紹與分析 37 3.1.2. 聲源定位演算法之實際應用 38 3.2. 感知計算(Awareness-Computation) 42 3.2.1. 基本概念介紹與分析 42 3.2.2. 感知計算與噪音抑制之應用 42 3.3. 整體架構之介紹與說明 50 第四章 聲源定位系統之硬體設計與規劃 52 4.1. 記憶體規劃 52 4.2. 交互相關性之硬體設計 54 4.3. 硬體設計流程 55 4.4. 硬體架構與FPGA結果 58 第五章 演算法的分析比較與結果 60 5.1. 乘法運算量之分析與比較 60 5.1.1. 交互相關性之乘法運算量分析與比較 60 5.1.2. 整體架構之乘法運算量比較 62 5.2. 效能分析與比較 64 5.2.1. MATLAB模擬之分析與比較 64 5.2.2. 實際環境的錄音實驗之分析與比較 74 第六章 結論與未來展望 83 參考文獻 84

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