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研究生： 陳奕彤
Chen, Yi-Tong
論文名稱： 陣列聲源於多點聲音傳遞效應分析之研究
Research on effect of multiple focusing points sound transmission by using a speaker array
指導教授： 涂季平
Too, Gee-Pinn
學位類別： 碩士
Master
系所名稱： 工學院 - 系統及船舶機電工程學系
Department of Systems and Naval Mechatronic Engineering
論文出版年： 2014
畢業學年度： 102
語文別： 英文
論文頁數： 77
中文關鍵詞： 集中聲音 、適應性時返法 、控制陣列聲源
外文關鍵詞： sound focusing, line array of controlled sources, adaptive time-reversal method
相關次數： 點閱：98  下載：6 
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    • 本論文係設計一陣列聲源控制系統達集中發聲之效果，其目的為增強特定區域之聲音訊號，並經由聲源控制輸入系統訊號抵減少其他特定區域之聲音相位。由適應性時間反轉法理論為基礎，利用Matlab程式得出陣列麥克風之權重函數，進行數值模擬於不同聲場邊界下之聲場分佈，並以Labview程式建立控制陣列聲源之實驗進行量測與驗證。

    為了印證此演算法之效應，本論文將各種不同參數條件做模擬分析，例如：改變聲源數目、接收端數目、與不同發射頻率之比較等；並在自由聲場與一般聲場環境下，比較受控制聲源之實驗結果，以量測不同聲場配置及驗證本研究之可行性。最後由一般聲場實驗中得知，聲音響亮區與安靜區可達5~12 dB值之差異。由此可知適應性時返法可實現一般聲場之能量集中，預計在未來可有多元化之運用。

    In this thesis, a system for focusing sound around desire locations by using speakers array of controlled sources is proposed. To increase acoustic signals in certain locations where the user is within but to reduce it in the other certain locations by controlling source signals is the main objective of this study. The program based on adaptive time-reversal theory is designed to obtain input weighting coefficients for speakers using Matlab. Experiments with speakers array of controlled sources and simulations under Matlab are established in order to observe the distribution of sound field under different boundary and control conditions.

    To test this system, a variety of simulation conditions are used, such as: the number of speakers array sources, the number of receivers, and emission source frequency. As for experiment, in the free field and reflected sound field under LabView are taken to compare results. Based on simulations and experiments, effects of controlled sound field are obtained. Finally, from general sound field experiments, the difference of sound pressure level between bright point and dark point is 5 ~ 12dB. Therefore, general sound field energy focusing is realized via present algorithm and expected to have a diversification of use in the future.

    Table of contents 中文摘要 I Abstract II 致謝 III Table of contents IV Table list VII Figure list IX I. Introduction 1 1.1 Research background and purpose 1 1.2 Literature Reviews 3 1.3 Research structure 6 1.4 Overview of the thesis 7 II. Basic theory 8 2.1 Basic theory of adaptive time-reversal for audio signal separation 8 2.2 Derivation of the weighting vectors of a multiple signal focusing algorithm 11 2.3 The algorithm to calculate the impulse response function 15 2.4 Signal processing of multiple signal focusing 17 III. Computer analysis of simulation modeling 18 3.1 Introduction of the simulation 18 3.1.1 The simulation process 18 3.2 Simulations for one reflective surface 22 3. 2.1 The modeling setting 22 3.2.2 The effect of the number of speakers array 23 3.2.3 Effect of the number of receivers 28 3.2.4 The effect of source frequency 33 3.3 Simulations for six reflective surfaces 38 3.3.1 The model setting 38 3.3.2 The effect of number of speakers 39 3.3.3 The effect of number of receivers 42 3.3.4 The effect of source frequency 45 IV. The experiment measurement and analysis for sound field focusing 49 4.1 The introduction of experiment devices 49 4.2 The experimental process 52 4.3 The experimental measurement for semi-anechoic room 53 4.3.1 The experiment set up 53 4.3.2 The experiment process 54 4.3.3 The experimental results 56 4.3.4 Comparison of the results of experiments and simulations 60 4.4 The experimental measurement in a reverberate sound field 62 4.4.1 The experimental setup 62 4.4.2 The experimental results 64 4.4.3 Comparison of the results of experiment and simulation 70 V. Conclusions and Future work 72 5.1 Conclusions 72 5.2 Future work 73 References 74

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