在惡劣的聲學條件下，例如嘈雜和混響環境中，語音品質容易下降。麥克風陣列訊號處理在惡劣聲學條件下，提高語音品質發揮著至關重要的作用。在這種條件下，聲源數目估計、語音增強和聲源分離特別具有挑戰性。在本論文中，我們提出了幾種基於陣列的技術來解決惡劣聲學環境中的聲源數目估計、語音增強和聲源分離問題。首先，我們專注於使用目標聲源到達方向(DOA)的先驗知識進行語音增強。實現了一種嵌入廣義旁辦消除(GSC)波束形成器結構的最小變異無失真響應，該結構具有串聯一個基於維納的後置濾波器(MVDR-GSC-PF)。MVDR-GSC-PF的最小變異無失真響應中的雜訊和混響共變異矩陣是使用加權預測誤差方法和阻塞矩陣技術進行估計。廣義旁辦消除波束形成器的自適應權重可用於減輕干擾訊號。廣義旁辦消除波束形成器的輸出端設計了基於維納的後置濾波器，以進一步減少殘餘雜訊和殘響。使用田口正交陣列設計的實驗表明，在語音質量感知評估(PESQ)、短時客觀清晰度(STOI)和訊號失真比(SDR)方面，MVDR-GSC-PF的性能優於基於阻塞的多通道维纳濾波器(BMWF)演算法、兩階段波束成形方法(TSBA)以及廣義旁辦消除和線性預測卡爾曼濾波器(ISCLP)演算法。對於聲源數目估計，我們也提出了一種基於使用連續阻塞過程的功率監測的巢式廣義旁辦消除(NGSC)方法。對於聲源分離，基於聲源數量和聲源到達方向（DOA）資訊提出了具有後置濾波器的線性限制最小變異後置濾波器（LCMV-PF）。自由場引導向量和透過巢式廣義旁辦消除估計的相對傳遞函數(RTF)向量之間的餘弦相似度可用於聲源到達方向估計。此外，將黃金分割搜尋（GSS）演算法結合餘弦相似度來加速聲源到達方向搜尋。透過結合遞迴平均和特徵值分解(EVD)的技術，LCMV-PF中的雜訊共變異數矩陣可以得到全秩重建。蒙特卡羅模擬和具有客觀品質測量的實驗用於評估所提出的巢式廣義旁辦消除和線性限制最小變異後置濾波器方法。

Speech quality is prone to degradation under adverse acoustic conditions, such as noisy and reverberant environments. Microphone array signal processing plays a vital role in improving speech quality under adverse acoustic conditions, where source counting, speech enhancement, and source separation can be particularly challenging. In this thesis, several array-based techniques are proposed to address the problems of source counting, speech enhancement, and source separation in adverse acoustic environments. First, we focus on speech enhancement with prior knowledge of the direction-of-arrival (DOA) of the target source. A minimum variance distortionless response beamformer embedded in a generalized sidelobe canceller (GSC) structure with a cascaded Wiener-based postfilter (MVDR-GSC-PF) is implemented. The noise and reverberation covariance matrices in MVDR-GSC-PF are estimated using the weighted prediction error method and a blocking matrix. Adaptive weighting of the GSC beamformer is used to mitigate the interfering source. A Wiener-based postfilter is cascaded at the output of the GSC beamformer to further reduce the residual noise and reverberation. Experiments designed using the Taguchi orthogonal arrays show that the MVDR-GSC-PF algorithm outperforms the blocking-based multichannel Wiener filter (BMWF) algorithm, the two-stage beamforming approach (TSBA), and the integrated sidelobe cancellation and linear prediction Kalman filter (ISCLP) algorithm in terms of the perceptual evaluation of speech quality (PESQ), the short-time objective intelligibility (STOI), and the signal-to-distortion ratio (SDR). For source counting, we also propose a nested GSC (NGSC) approach based on power monitoring using a successive blocking procedure. For source separation, a linearly constrained minimum variance beamformer with postfiltering (LCMV-PF) is proposed based on the number of sources and the DOA information. The cosine similarity between the free-field steering vector and the vector of the relative transfer function (RTF) via NGSC is employed for DOA estimation. The golden section search (GSS) algorithm is applied with cosine similarity to accelerate the DOA search. By combining the techniques of recursive averaging and eigenvalue decomposition (EVD), the noise covariance matrix in LCMV can be reconstructed with full rank. Monte Carlo simulations and experiments with objective quality measures are used to evaluate the proposed NGSC and LCMV-PF methods.

致謝 I
摘要 II
ABSTRACT IV
TABLE OF CONTENTS VI
LIST OF TABLES IX
LIST OF FIGURES XI
CHAPTER 1. INTRODUCTION 1
CHAPTER 2. CONVENTIONAL ARRAY SIGNAL PROCESSING APPROACHES 7
2.1 Array signal model and microphone array system 7
2.2 Source counting algorithms 9
2.2.1 Delay-and-sum (DAS) beamforming approach 9
2.2.2 Minimum power distortionless response (MPDR) approach 13
2.2.3 Multiple signal classification (MUSIC) algorithm 15
2.2.4 Eigenvalue decomposition (EVD)-based source counting algorithm 17
2.2.5 Minimum description length (MDL) source counting algorithm 18
2.2.6 Second order statistic of the eigenvalue (SORTE) source counting algorithm 21
2.2.7 Multistage Wiener filter (MSWF) source counting algorithm 22
2.3 Source localization algorithms 25
2.4 Source separation algorithms 28
2.4.1 Superdirective beamforming (SBF) 28
2.4.2 Linearly constrained minimum power (LCMP) 29
2.5 Simulations and experiments 31
2.5.1 Simulation and experimental settings 31
2.5.2 Source counting results 36
2.5.3 Source localization results 43
2.5.4 Source separation results 48
CHAPTER 3. SPEECH ENHANCEMENT WITH MULTICHANNEL MICROPHONES 50
3.1 Blocking-based multichannel Wiener filter 50
3.2 Two-stage beamforming approach for denoising and dereverberation 53
3.3 Integrated sidelobe cancellation and linear prediction Kalman filter (ISCLP) 57
3.4 Proposed MVDR-GSC-PF beamformer 60
3.5 Simulations and experiments 67
3.5.1 Simulation settings 67
3.5.2 Simulation results 69
3.5.3 Experimental results from actual room recordings 72
CHAPTER 4. PROPOSED NESTED GSC-BASED (NGSC) STRUCTURE FOR SOURCE COUNTING, LOCALIZATION, AND SEPARATION 75
4.1 NGSC source counting algorithm 75
4.2 NGSC source localization algorithm 82
4.3 LCMV-PF source separation algorithm 83
4.4 Simulation and experimental results 86
4.4.1 Settings 86
4.4.2 Source counting results 87
4.4.2.1 Analysis of the MSWF-based source counting limitation 97
4.4.2.2 Analysis of the NGSC threshold 98
4.4.3 Source localization results 101
4.4.3.1 Analysis of the NGSC source counting and localization algorithm for the case of sources 105
4.4.4 Source separation results 107
CHAPTER 5. CONCLUSIONS AND FUTURE WORK 109
5.1 Conclusions 109
5.2 Future work 111
REFERENCES 112
ABBREVIATIONS 125
PUBLICATIONS 127

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