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一种适用于混响环境的双传声器自适应指向性算法 被引量:1

An Adaptive Dual-microphone Directional Algorithm for Reverberant Environment
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摘要 在许多语音信号处理系统(如助听器)中,都采用了指向性算法处理在空间上相互分离的信号,但在客厅、会议室等室内环境中混响的存在严重影响了指向性系统的降噪性能,现有的去混响算法不能有效地抑制干扰噪声。该文提出一种适用于混响环境的自适应双传声器指向性算法,采用两个间距很小的全向性传声器,将自适应零限波束形成(ANF)结构和利用概率模型抑制混响的方法相结合,实现了在混响环境中的自适应指向性。与现有指向性和抑制混响算法相比,该算法采用一个简单的结构同步实现了指向性和抑制混响,具有较低的复杂度和较强的实时性。仿真验证了算法在混响环境中的指向性降噪性能。 In many acoustic signal processing systems(e.g.hearing aids),directional algorithm is adopted to process signals from separated sources.However,reverberation in a living room or a conference room usually degrades noise reduction of a directional system,while the existing dereverberation algorithms can not effectively suppress interference noises.In this paper,an adaptive dual-microphone directional algorithm with two closely arranged omnidirectional microphones is proposed for reverberant environment.The proposed algorhthm combines the Adaptive Null-Forming(ANF) structure and a dereverberation algorithm based on statistical model to achieve adaptive directionality in reverberant environment.Compared with existing directional or dereverbertion algorithms,the proposed algorithm uses a simple structure to realize directionality and dereverberation synchronically,as well as owns low complexity and real-time capability.Finally,the directional performance of the proposed algorithm in reverberant enviroment was verified by simulation results.
出处 《电子与信息学报》 EI CSCD 北大核心 2011年第11期2652-2657,共6页 Journal of Electronics & Information Technology
基金 国家自然科学基金(60970136) 中国科学院"科技助残行动计划"(KGCX-YW-616)资助课题
关键词 双传声器 混响 自适应 指向性 概率模型 Dual-microphone Reverberation Adaptive Directional Statistical model
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  • 1Griffiths L J and Jim C W. An alternative approach to linearly constrained adaptive beamforming[J]. IEEETransactions on Antennas and Propagation, 1982, 30(1): 27-34.
  • 2Luo Fa-long and Yang Jun. Adaptive null-forming scheme in digital hearing aids [J]. IEEE Transactions on Signal Processing, 2002, 50(7): 1583-1590.
  • 3Cox H, Zeskind R, and Owen M. Robust adaptive beamforming[J]. IEEE Transactions on Acoustics, Speech, Signal Processing, 1987, 35(10): 1365-1376.
  • 4Kokkinakis K and Loizou P C. Selective-tap blind dereverberation for two-microphone enhancement of reverberant speech[J]. IEEE Signal Processing Letters, 2009, 16(11): 961-964.
  • 5Kumar K and Stern R M. Maximum-likelihood-based cepstral inverse filtering for blind speech dereverberation[C]. International Conference on Acoustics Speech and Signal Processing (ICASSP), Dallas, Texas, USA, March 2010: 4282-4285.
  • 6Doclo S. Multi-microphone noise reduction and dereverberation techniques for speech applications[D]. [Ph.D. dissertation], Katholieke Universiteit Leuven, 2003.
  • 7Yegnanarayana B and Satayanarayana P. Enhancement of reverberant speech using LP residual signal[J]. IEEE Transactions on Speech and Audio Processing, 2000, 8(3): 267-281.
  • 8Wu M and Wang D L. A two-stage algorithm for one-microphone reverberant speech enhancement[J]. IEEE Transactions on Audio, Speech, Lang Process, 2006, 14(3): 774-784.
  • 9Habets E A P and Gannot S. Dual-microphone speech dereverberation using a reference signal [C]. InternationalConference on Acoustics Speech and Signal Processing (ICASSP), Hawaii, USA, April 2007, (4): 901-904.
  • 10Lsllmann H W and Vary P. Low delay noise reduction and dereverberation for hearing aids[J]. EURASIP Journal on Advances in Siqnal Processinq, 2009, 1(1): 1-9.

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