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作者(中文):花逸芯
作者(外文):Hua, Yi-Hsin
論文名稱(中文):整合式空間聲場錄音及回放陣列系統
論文名稱(外文):An Integrated Recording-Rendering Array System for Spatial Sound Fields
指導教授(中文):白明憲
指導教授(外文):Bai, Mingsian R.
口試委員(中文):李昇憲
洪健中
口試委員(外文):Sheng-Shian Li
Chien-Chong Hong
學位類別:碩士
校院名稱:國立清華大學
系所名稱:動力機械工程學系
學號:101033529
出版年(民國):103
畢業學年度:102
語文別:英文中文
論文頁數:60
中文關鍵詞:聲場重建陣列系統
外文關鍵詞:sound field reproductionarray system
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本文提出一個可以用來錄製空間聲場以及回放的整合式陣列系統。此架構類似於離散時間訊號處理的分析以及合成濾波器系統,此陣列系統的運作包括兩個階段:在第一個階段,利用麥克風陣列分析以及將聲場「編碼」,將聲場簡化為僅由聲源方向以及訊號兩個物理量來表示,此麥克風陣列是利用多重訊號分類(Multiple Signal Classification)這種常用的到達方向(Direction of Arrival)方法去估計聲源的方位,並將各種不同的聲源訊號分離。第二個階段則是一個喇叭陣列,它可以「解碼」以及重現在第一階段中獲得的聲場資訊,其中包含聲源方位、訊號,在這個階段我們透過在重建範圍內設置大量的假想控制點,來對欲重建的聲場做取樣,進而由這些控制點來反推喇叭訊號。而實際上在重現聲場的時候,可能會因為環境反射的因素而降低喇叭內部重建區域的音質,因此本文也討論,如何同時有效地控制喇叭外部聲場的輻射,並維持重建聲場的品質。過程中的逆矩陣運算則主要是由截斷奇異值(Truncated Singular Value Decomposition)和最佳化(Convex Optimization)兩種方法求解。最後,本文也提出客觀與主觀的評估測試,來衡量此陣列系統對於聲場的分離以及重建之效果。
An integrated recording and reproduction array system for spatial audio is presented. The system has a generic framework akin to the analysis-synthesis filterbanks in discrete time signal processing. The architecture of the system consists of two stages. In the first stage, an analysis microphone array that “encodes” the sound field into sparse source components in the angle space. Direction of Arrival (DOA) of sound sources is estimated by MUltiple SIgnal Classification (MUSIC). The second stage is a field synthesis by loudspeaker array that “decodes”, or reproduces, the sound field on the basis of the source components obtained in the first stage. Specifically, this stage can be formulated as a deconvolution problem by matching the reproduced and target fields at numerous control points in the interior domain of the loudspeaker array, while minimizing the external radiation to mitigate the adverse effects of boundary reflections. The inverse problem is solved by Truncated Singular Value Decomposition (TSVD) or Convex Optimization (CVX) algorithms. Since the performance of sound field synthesis may be constrained by aliasing frequency, a hybrid method combing deconvolution and vector panning is proposed here for improving the reproduction. The result of source extraction and the performance of reproduction are also evaluated.
摘要 I
Abstract II
誌謝 III
Table of Contents IV
TABLE LIST V
FIGURE LIST VI
Chapter 1 INTRODUCTION 1
Chapter 2 SOUND FIELD ANALYSIS USING A MICROPHONE ARRAY 6
2.1 Source tracking 7
2.2 Source Extraction 8
Chapter 3 SOUND FIELD SYNTHESIS USING A LOUDSPEAKER ARRAY 11
3.1 Deconvolution-based sound field reproduction 12
3.2 Minimizing external radiation 14
3.3 Panning-based sound field reproduction 14
3.4 Hybrid approach 15
3.5 Integrated microphone and loudspeaker array 16
Chapter 4 NUMERICAL SIMULATIONS 17
4.1 Plane-wave decomposition using a microphone array 18
4.2 Reproduction by pressure matching 18
4.3 Binaural simulation for the synthesized sound field 20
4.4 Dark-zone control 21
Chapter 5 EXPERIMENTAL PERFORMANCE EVALUATIONS 23
5.1 Plane-wave decomposition using a microphone array 24
5.2 Performance evaluation of the loudspeaker array reproduction 24
Chapter 6 CONCLUSIONS 27
REFERENCES 29
APPENDIX 33
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