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研究生: 王基鴻
Ji-hong Wang
論文名稱: 空間濾波器於麥克風陣列之設計
Beamforming Design of Microphone array
指導教授: 徐國鎧
Kuo-Kai Shyu
口試委員:
學位類別: 碩士
Master
系所名稱: 資訊電機學院 - 電機工程學系
Department of Electrical Engineering
畢業學年度: 100
語文別: 中文
論文頁數: 110
中文關鍵詞: 空間濾波器線性最小變異限制麥克風陣列相對延遲時間傳遞衰減係數到達方向
外文關鍵詞: linear constrained minimum variance (LCMV), microphone array, channel attenuation factor, direction of arrival (DOA), time difference of arrival (TDOA)
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  •   本論文使用空間濾波器來處理麥克風之均勻線性陣列與均勻圓形陣列的訊號處理,應用線性限制最小變異(LCMV)演算法,目的為增強目標方向訊號並壓抑非目標方向訊號,並藉由估測聲源到各顆麥克風之間的傳遞衰減係數,調整麥克風電路的放大倍率,以降低麥克風之間的差異性。接著利用麥克風陣列訊號的空間資訊做聲源到達方向(DOA)估測,得到空間濾波器輸入端對於聲源方向的相對延遲時間(TDOA)。最後設計麥克風陣列類比放大電路接收聲音訊號,以驗證空間濾波器系統,並探求均勻線性陣列與均勻圓形陣列之異同。


    This thesis investigates the signal processing of a microphone array when the arrangement of the microphones is a uniform linear array (ULA) or a uniform circular array (UCA), designing an implementation of a microphone-array beamforming. In real world environment, the speech signal received by a set of microphones contains the desired signal, interference, and ambient noise. In order to enhance the desired signal and reduce other signal, the algorithm, linearly constrained minimum variance (LCMV), is used. By estimating the channel attenuation factor between the source signal and each microphone, one can adjust the gain of the circuit to decline the difference between microphones. After that, the estimate of direction of arrival (DOA) from the spatial information of the microphone-array signal, the time difference of arrival (TDOA) between microphones due to the direction of source is obtained. Finally, by designing a microphone-array circuit to receive the voice signal, this study verifies the beamforming system in a noisy speech environment, and then discusses the difference between the signals received by a set of ULA and UCA.

    摘要 i Abstract ii 誌謝 iii 目錄 iv 圖目錄 vi 表目錄 xi 第一章  緒論 1 1.1 研究動機 1 1.2 研究目標 3 1.3 論文架構 4 第二章  空間濾波器 5 2.1 空間濾波器簡介 5 2.2 延遲相加空間濾波器 8 2.3 空間響應 13 2.4 線性限制最小變異空間濾波器 26 2.5 空間濾波器之模擬 35 2.6 傳遞衰減係數估測 48 第三章  聲源訊號到達角估測 50 3.1 前言 50 3.2 聲音訊號模型 50 3.3 聲源到達角估測 52 3.4 聲源到達角估測模擬 57 第四章  實驗與討論 65 4.1 系統架構設計 65 4.2 實驗評量方式 67 4.3 實驗平台 70 4.4 實驗結果與討論 72 第五章  結論與未來展望 91 參考文獻 92

    [1] H. Krim and M. Viberg, “Two Decades of Array Signal Processing Research,” IEEE Signal Processing Magazine, July 1996.
    [2] P. Stoica and A. Nehorai, “Performance Study of Conditional and Unconditional Direction-of-Arrival Estimation,” IEEE Trans. Acoustics Speech Signal Process., vol. 38, pp. 1783–1795, Oct. 1990.
    [3] B. D. Van Veen and K. M. Buckley, “Beamforming: A Versatile Approach to Spatial Filtering,” IEEE ASSP Magazine, pp. 4–24, April 1988.
    [4] W. Herbordt and W. Kellermann, “Adaptive beamforming for audio signal acquisition,” in Adaptive Signal Processing: Applications to Real-World Problems, J. Benesty and Y. Huang, eds., Berlin, Germany: Springer-Verlag, 2003.
    [5] R. T. Compton, Jr., Adaptive Antennas: Concepts and Performance. Englewood Cliffs, NJ: Prentice-Hall, 1988.
    [6] J. Benesty, M. M. Sondhi, and Y. Huang, eds., Springer Handbook of Speech Processing. Berlin, Germany: Springer-Verlag, 2007.
    [7] M. Brandstein, and D. Ward, eds., Microphone Arrays: Signal Processing Techniques and Applications, Berlin, Germany: Springer-Verlag, 2001.
    [8] D. E. Dudgeon, “Fundamentals of digital array processing,” Proc. IEEE, vol. 65, pp. 898–904, June 1977.
    [9] O. L. Frost, III, “An algorithm for linearly constrained adaptive array processing,” Proc. IEEE, vol. 60, pp. 926–935, Aug. 1972.
    [10] H. Gish and D. Cochran, “Generalized coherence,” in Proc. IEEE Int. Conf. Acoustic Speech, Signal Processing, vol. 5, pp. 2745–2748, 1988.
    [11] J. P. Dmochowski, J. Benesty, and S. Affes, “Direction of arrival estimation using the parameterized spatial correlation matrix,” IEEE Trans. Audio, Speech, Language Process., vol. 15, pp. 1327–1339, May 2007.
    [12] M. Souden, J. Benesty, and S. Affes, “Broadband Source Localization From an Eigenanalysis Perspective,” IEEE Trans. Audio, Speech, Language Process., vol. 15, pp. 1575–1587, Aug. 2010.
    [13] J. Benesty, J. Chen, and Y. Huang, “Time-delay estimation via linear interpolation and cross-correlation,” IEEE Trans. Speech Audio Process., vol. 12, pp. 509–519, Sept. 2004.
    [14] J. Dmochowski, J. Benesty, and S. Affes, “On Spatial Aliasing in Microphone Arrays,” IEEE Trans Signal Process., vol. 57, pp. 1383–1395, April 2009.
    [15] NOISE-92(1993) in http://spib.rice.edu/spib/select_noise.html.
    [16] Wimston E. Kock著,張丹 譯,“波光與波聲”,臺灣商務印書館,1983.
    [17] Y. Huang and J. Benesty, “Adaptive multi-channel least mean square and Newton algorithms for blind channel identification,” Signal Process., vol. 82, pp. 1127–1138, Aug. 2002.
    [18] Y. Huang and J. Benesty, “A class of frequency-domain adaptive approaches to blind multi-channel identification,” IEEE Trans. Signal Process., vol. 51, pp. 11–24, Jan. 2003.
    [19] J. Chen, J. Benesty, and Y. Huang, “Robust time delay estimation exploiting redundancy among multiple microphones,” IEEE Trans. Speech Audio Process., vol. 11, pp. 549–557, Nov. 2003.
    [20] H. L. Van Trees, Optimum Array Processing. Part IV of Detection, Estimation, and Modulation Theory. New York: John Wiley & Sons, Inc., 2002.
    [21] K. Kobayashi, Y. Haneda, K. Furuya, and A. Kataoka, “A Hands-Free Unit with noise Reduction by Using Adaptive Beamformer,” IEEE Trans. Consumer Electronics, vol. 54, pp. 116–122, Feb. 2008.0000
    [22] X.Zhang and J. H. L. Hansen, “CSA-BF: a constrained switched adaptive beamformer for speech enhancement and recognition in real car environments,” IEEE Trans. Speech Audio Process., vol. 11, pp. 733–745, Nov. 2003.
    [23] D. H. Johnson and D. E. Dudgeon, Array Signal Processing – Concepts and Techniques. Englewood Cliffs, NJ: Prentice-Hall, 1993.
    [24] J. L. Flanagan, J. D. Johnson, R. Zahn, and G. W. Elko, “Computer-steered microphone arrays for sound transduction in large rooms,” J. Acoust. Soc. Amer., vol. 75, pp. 1508–1518, Nov. 1985.

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