[1] BRANDSTEIN M, WARD D. Microphone arrays: signal processing techniques and applications [M]. Berlin: Springer Verlag, 2001: 116, 229-245.
[2] GRIFFITHS L, JIM C. An alternative approach to linearly constrained adaptive beamforming [J]. IEEE Transactions on Antennas and Propagation, 1982, 30(1): 27-34.
[3] KRUEGER A, WARSITZ E, Haeb-Umbach R. Speech enhancement with a GSC-Like structure employing eigenvector-based transfer function ratios estimation [J]. IEEE Transactions on Audio, Speech, and Language Processing, 2011, 19(1): 206 -219.
[4] 欧世峰,赵晓晖,顾海军.改进的基于信号子空间的多通道语音增强算法[J].电子学报,2005, 33(10): 1786-1789.
OU Shi-feng, ZHAO Xiao-hui, GU Hai-jun. An improved speech enhancement approach based on signal subspace with multi-Input [J]. ACTA ELECTRONICA SINICA, 2005, 33(10): 1786-1789.
[5] ELKO G W, PONG A N. A simple adaptive first-order differential microphone[C]∥Proceedings of IEEE International Conference on Applications of Signal Processing to Audio and Acoustics. New York: IEEE, 1995: 169-172.
[6] BUCK M, WOLFF T, HAULICK T, et al. A compact microphone array system with spatial post-filtering for auto-motive applications[C]∥ Proceedings of IEEE International Conference on Acoustics, Speech and Signal Processing. Taipe: IEEE, 2009: 221-224.
[7] CHEN J, PHUA K, SHUEA L, et al. Performance evalua-tion of adaptive dual microphone system [J]. Speech Commu-nication, 2009, 51(12): 1180-1193.
[8] NINO-DE-RIVERA L, PEREZ-MEANA H, SANCHEZ-SINENCIO E. A modular analog NLMS structure for system identification[C]∥Proceedings of the 40th Midwest Sym-posium on Circuits and Systems. Sacramento : IEEE, 1997: 835-840.
[9] 王振力,张雄伟, 杨吉斌.一种新的快速自适应滤波算法的研究[J].通信学报,2005, 26(11): 1-6.
WANG Zhen-li, ZHANG Xiong-wei YANG Ji-bin,et al Study of a new fast adaptive filtering algorithm [J]. Journal on Communications, 2005, 26(11) : 1-6.
[10] YASUKAWA H, SHIMADA S. An Acoustic echo can-celler using subband sampling and decorrelation methods [J]. IEEE Transactions on Signal Processing, 1993, 41(2): 926-930.
[11] FERRARA E. Fast implementations of LMS adaptive filters[J]. IEEE Transactions on Acoustics, Speech and Signal Processing, 1980, 28(4): 474-475.
[12] SULYMAN A I, ZERGUINE A. Convergence and steady-state analysis of a variable step-size NLMS algorithm [J]. IEEE Transactions on Signal Processing, 2003, 83(6): 1255-1273.
[13] BOLL S. Suppression of acoustic noise in speech using spectral subtraction[J]. IEEE Transactions on Acoustics, Speech and Signal Processing, 1979, 27(2): 113-120.
[14] SOHN J, KIM N S, SUNG W. A statistical model-based voice activity detection [J]. IEEE Signal Processing Letters, 1999, 6 (1): 13.
[15] MARTIN R. Noise power spectral density estimation based on optimal smoothing and minimum statistics [J]. IEEE Transactions on Speech and Audio Processing, 2001, 9(5): 504-512.
[16] MARTIN R. Bias compensation methods for minimum statistics noise power spectral density estimation [J], Signal Processing, 2006, 86(6): 1215-1229. |