Speech dereverberation algorithm using transfer function estimates with overestimated order

Author:

Hikichi Takafumi1,Delcroix Marc12,Miyoshi Masato12

Affiliation:

1. NTT Communication Science Laboratories, NTT Corporation

2. Graduate School of Information Science and Technology, Hokkaido University

Publisher

Acoustical Society of Japan

Subject

Acoustics and Ultrasonics

Reference19 articles.

1. 1) X. Sun and S. C. Douglas, “A natural gradient convolutive blind source separation algorithm for speech mixtures,” Proc. ICA, pp. 59–64 (2001).

2. 2) S. Amari, S. C. Douglas, A. Cichocki and H. H. Yang, “Multichannel blind deconvolution and equalization using the natural gradient,” Proc. IEEE Workshop on Signal Processing in Advances in Wireless Communications, pp. 101–104 (1997).

3. 3) T. Nakatani and M. Miyoshi, “Blind dereverberation of single channel speech signal based on harmonic structure,” Proc. ICASSP IEEE, Vol. 1, pp. 92–95 (2003).

4. 4) T. Nakatani, K. Kinoshita, M. Miyoshi and P. Zolfaghari, “Harmonicity based monaural speech dereveberation with time warping and F0 adaptive window,” Proc. INTERSPEECH ICSLP, Vol. 2, pp. 873–876 (2004).

5. Subspace methods for the blind identification of multichannel FIR filters

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1. Variance Compensation for Recognition of Reverberant Speech with Dereverberation Preprocessing;Robust Speech Recognition of Uncertain or Missing Data;2011

2. Multi-microphone Speech Dereverberation Using Eigen-decomposition;Speech Dereverberation;2010

3. Static and Dynamic Variance Compensation for Recognition of Reverberant Speech With Dereverberation Preprocessing;IEEE Transactions on Audio, Speech, and Language Processing;2009-02

4. Calculating Inverse Filters for Speech Dereverberation;IEICE Transactions on Fundamentals of Electronics, Communications and Computer Sciences;2008-06-01

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