博碩士論文 985201092 詳細資訊




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姓名 吳孟儒(Meng-ju Wu)  查詢紙本館藏   畢業系所 電機工程學系
論文名稱 自適應性麥克風陣列空間濾波器設計與實現
(Design and Implementation ofAdaptive Microphone-Array Beamforming)
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摘要(中) 本論文針對線性均勻的麥克風陣列訊號處理設計一自適應性空間濾波器。目的為增強期望目標訊號方向的聲音,以及降低其他方向的干擾與雜訊。本研究以線性限制最小變異(LCMV)演算法應用於增強期望目標訊號方向的聲音訊號,加上估測聲音訊號到達麥克風陣列各顆麥克風的傳遞衰減係數,藉此來調整陣列各顆麥克風放大電路的倍率,以降低各顆麥克風間的差異性,讓空間濾波器的輸出較理想,接著,再使用多通道互相關係數(MCCC)演算法估測聲音的到達方向(DOA),以此得到空間濾波器輸入訊號之最佳延遲時間,最後研製麥克風陣列類比放大電路接收聲音訊號,實現及驗證所設計的空間濾波器系統。
摘要(英) This thesis investigates the signal processing of a uniform linear microphone array to design and implement an adaptive microphone-array beamforming. In practical world environments, the signal captured by a set of microphones in a speech communication system is a signal mixed with the desired signal, interference, and ambient noise. A promising solution of proper speech acquisition with reduced noise and interference in this context consists in using the linearly constrained minimum variance (LCMV) beamformining to reject the interference, reduce the overall mixture energy, and preserve the target signal. This approach requires such knowledge as the direction of arrival (DOA); therefore an estimator based on the multichannel cross correlation coefficient (MCCC) is also used. In addition, an eigenanalysis of the parameterized spatial correlation matrix is performed and reveals that such analysis allows one to estimate the channel attenuation from factors such as uncalibrated microphones. This estimate generalizes the broadband minimum variance spatial spectral estimator to more general signal models. Finally, experimental results show that the developed microphone array amplifier circuit and accompanied with signal processing algorithms successfully improve the target signal in the noisy environment.
關鍵字(中) ★ 多通道互相關係數(MCCC)
★ 傳遞衰減係數
★ 麥克風陣列
★ 束波成形
★ 到達方向(DOA)
★ 空間濾波器
★ 線性限制最小變異(LCMV)
關鍵字(英) ★ spatial filter
★ Linearly constrained minimum variance (LCMV)
★ beamforming
★ microphone-array
★ channel attenuation from factors
★ direction of arrival (DOA)
★ multichannel cross-correlation coefficient (MCCC
論文目次 摘要...........................................I
Abstract......................................II
目錄..........................................IV
圖目錄........................................VI
表目錄.......................................XII
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.........................6
2.3 空間響應...................................9
第三章適應性濾波..............................15
3.1適應性濾波器簡介...........................15
3.2 線性限制最小變異空間濾波器................16
3.3 線性限制最小變異濾波器之模擬..............22
3.4傳遞衰減係數估測...........................27
第四章信號到達方向估測........................29
4.1 基本介紹..................................29
4.2聲音訊號模型...............................30
4.3向前空間線性預估法.........................36
第五章實驗與討論..............................42
5.1 系統架構設計..............................42
5.2 實驗評量方法..............................45
5.3 實驗平台..................................50
5.4 實驗結果與討論............................56
第六章結論與未來展望..........................73
參考文獻......................................74
參考文獻 [1] Alan V. Oppenheim, Ronald W. Schafer, Discrete-Time Signal Processing, Prentice Hall, 1989.
[2] D. Johnson and D. Dudgeon, Array Signal Processing:Concepts and Techniques, Prentice Hall, Englewood Cliffs, New Jersey, 1993
[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] D. Van Compernolle, “Switching adaptive filters for enhancing noisy and reverberant speech from microphone array recordings,” in Proc. IEEE ICASSP, pp. 833–836,1990.
[6] R. T. Compton, Jr., Adaptive Antennas: Concepts and Performance. Englewood Cliffs, NJ: Prentice-Hall, 1988.
[7] J. Benesty, M. M. Sondhi, and Y. Huang, eds., Springer Handbook of Speech Processing. Berlin, Germany: Springer-Verlag, 2007.
[8] M. Branstein and D. B. Ward, Eds., Microphone Arrays: Signal Processing Techniques and Applications. Berlin, Germany: Springer,2001.
[9] J. Benesty and Y. Huang, Eds., Adaptive Signal Processing: Applications to Real-World Problems. Berlin, Germany: Springer, 2003.
[10] S. A. Schelkunoff, “A mathematical theory of linear arrays,” Bell Syst. Tech. J., vol. 22, pp. 80–107, Jan. 1943.
[11] M. Brandstein and D. B. Ward, eds., Microphone Arrays: Signal Processing Techniques and Applications. Berlin, Germany: Springer-Verlag, 2001.
[12] D. E. Dudgeon, “Fundamentals of digital array porcessing,” Proc. IEEE, vol.65, pp. 898–904, June 1977.
[13] 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.
[14] J. L. Flanagan, D. A. Berkley, G. W. Elko, J. E. West, and M. M. Sondhi, “Autodirective microphone systems,” Acusti., vol. 73, pp. 58–71, Feb. 1991.
[15] D. H. Johnson and D. E. Dudgeon, Array Signal Processing–Concepts and Techniques. Englewood Cliffs, NJ: Prentice-Hall, 1993.
[16] hed adaptive beamformer for speech enhancement and recognition in real car environments,” IEEE Trans. Speech Audio Process., vol. 11, pp. 733–745, Nov. 2003.
[17] H. L. Van Trees, Optimum Array Processing. Part IV of Detection, Estimation, and Modulation Theory. New York: John Wiley & Sons, Inc., 2002.
[18] B. G. Bardsley and D. A. Christensen, “Beam pattern from pulsed ultrasonic transducers using linear systems theory,” J. Acoust. Soc. Am., vol. 69, pp. 25–30, Jan. 1981.
[19] J. W. Goodman, Introduction to Fourier Optics. New York: McGraw-Hill, 1968.
[20] O. L. Frost, III, “An algorithm for linearly constrained adaptive array processing,” Proc. IEEE, vol. 60, pp. 926–935, Aug. 1972.
[21] J. Chen, J. Benesty, and Y. Huang, “Time delay estimation in room acoustic environments: an overview,” EURASIP J. Applied Signal Process., vol. 2006,Article ID 26503, 19 pages, 2006.
[22] 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.
[23] NOISE-92 (1993) in http://spib.rice.edu/spib/select_noise.html .
[24] J. DiBiase, H. Silverman, and M. Brandstein, “Robust localization in reverberant rooms,” in Microphone Arrays: Signal Processing Techniques and Applications, M. Branstein and D. Ward, eds., Berlin, Germany: Springer, 2001.
[25] S. M. Griebel and M. S. Brandstein, “Microphone array source localization using realizable delay vectors,” in Proc. IEEE WASPAA, pp. 71–74, 2001.
[26] A. Hyv ̈arinen, J. Karhunen, and E. Oja, Independent Component Analysis. London, England: John Wiley & Sons, 2001.
[27] 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, Sep. 2004.
[28] delay estimation exploiting redundancy among multiple microphones,” IEEE Trans. Speech Audio Process., vol. 11, pp. 549–557, Nov. 2003.
[29] Haykin, S. Modern Filters, Macmillan, New York. 1989.
[30] R. L. Pritchard, “Maximum directivity index of a linear point array,” J. Acoust. Soc. Am.vol. 26, Issue 6, pp.1034-1039,1954
[31] Wimston E. Kock 著,張丹 譯,“波光與波聲”,臺灣商務印書館,1983
[32] Kenneth Boyce,“提升行動通訊語音品質遠場雜訊抑制麥克風陣列出線”,新通訊 2009 年 8 月號 102 期《 技術前瞻 》
[33] 王小川編著,“語音訊號處理”,全華科技,2002
[34] J. J. Groen, “Social hearing handicap: its measurement by speech audiometry in noise,” J. Acoust. Soc. Am., vol. 8, pp. 182-183. 1969.
[35] R. Plomp, “Auditory handicap of hearing impairment and the limited benefit of hearing aids, ” J. Acoust. Soc. Am. ,vol. 63, pp. 533-549. 1978.
[36] R. Plomp, A. M. Mimpen, “Speech-reception threshold for sentences as a function of age and noise level, ” J. Acoust. Soc. Am., vol. 66, pp. 1333-1342. 1979.
指導教授 徐國鎧(Kuo-kai Shyu) 審核日期 2011-8-2
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