Speech Dereverberation Based on Maximum-Likelihood Estimation With Time-Varying Gaussian Source Model
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Biing-Hwang Juang | Tomohiro Nakatani | Takuya Yoshioka | Marc Delcroix | Keisuke Kinoshita | Masato Miyoshi | B. Juang | T. Nakatani | K. Kinoshita | M. Miyoshi | Marc Delcroix | Takuya Yoshioka
[1] Bayya Yegnanarayana,et al. Enhancement of reverberant speech using LP residual signal , 2000, IEEE Trans. Speech Audio Process..
[2] Philippe Loubaton,et al. Prediction error method for second-order blind identification , 1997, IEEE Trans. Signal Process..
[3] Biing-Hwang Juang,et al. Robust blind dereverberation of speech signals based on characteristics of short-time speech segments , 2007, 2007 IEEE International Symposium on Circuits and Systems.
[4] Masato Miyoshi. ESTIMATING AR PARAMETER-SETS FOR LINEAR-RECURRENT SIGNALS IN CONVOLUTIVE MIXTURES , 2003 .
[5] Yunxin Zhao,et al. An EM algorithm for linear distortion channel estimation based on observations from a mixture of Gaussian sources , 1999, IEEE Trans. Speech Audio Process..
[6] Chrysostomos L. Nikias,et al. EVAM: an eigenvector-based algorithm for multichannel blind deconvolution of input colored signals , 1995, IEEE Trans. Signal Process..
[7] Hagai Attias,et al. An EM Method for Spatio-temporal Blind Source Separation Using an AR-MOG Source Model , 2006, ICA.
[8] Dirk T. M. Slock,et al. Multivariate LP Based MMSE-ZF Equalizer Design Considerations and Application to Multimicrophone Dereverberation , 2007, 2007 IEEE International Conference on Acoustics, Speech and Signal Processing - ICASSP '07.
[9] Masato Miyoshi,et al. Inverse filtering of room acoustics , 1988, IEEE Trans. Acoust. Speech Signal Process..
[10] J. Flanagan,et al. Computer‐steered microphone arrays for sound transduction in large rooms , 1985 .
[11] Dirk T. M. Slock,et al. Blind fractionally-spaced equalization, perfect-reconstruction filter banks and multichannel linear prediction , 1994, Proceedings of ICASSP '94. IEEE International Conference on Acoustics, Speech and Signal Processing.
[12] Tomohiro Nakatani,et al. Maximum likelihood approach to speech enhancement for noisy reverberant signals , 2008, 2008 IEEE International Conference on Acoustics, Speech and Signal Processing.
[13] Andrew R. Barron,et al. Mixture Density Estimation , 1999, NIPS.
[14] J. Cardoso,et al. Maximum likelihood approach for blind audio source separation using time-frequency Gaussian source models , 2005, IEEE Workshop on Applications of Signal Processing to Audio and Acoustics, 2005..
[15] Marc Delcroix,et al. Inverse Filtering for Speech Dereverberation Less Sensitive to Noise and Room Transfer Function Fluctuations , 2007, EURASIP J. Adv. Signal Process..
[16] Jacob Benesty,et al. Speech Acquisition and Enhancement in a Reverberant, Cocktail-Party-Like Environment , 2006, 2006 IEEE International Conference on Acoustics Speech and Signal Processing Proceedings.
[17] Biing-Hwang Juang,et al. Fundamentals of speech recognition , 1993, Prentice Hall signal processing series.
[18] 古井 貞煕,et al. Digital speech processing, synthesis, and recognition , 1989 .
[19] Klaus Uwe Simmer,et al. Superdirective Microphone Arrays , 2001, Microphone Arrays.
[20] Henrique S. Malvar,et al. Speech dereverberation via maximum-kurtosis subband adaptive filtering , 2001, 2001 IEEE International Conference on Acoustics, Speech, and Signal Processing. Proceedings (Cat. No.01CH37221).
[21] Biing-Hwang Juang,et al. Study on Speech Dereverberation with Autocorrelation Codebook , 2007, 2007 IEEE International Conference on Acoustics, Speech and Signal Processing - ICASSP '07.
[22] Tomohiro Nakatani,et al. Importance of Energy and Spectral Features in Gaussian Source Model for Speech Dereverberation , 2007, 2007 IEEE Workshop on Applications of Signal Processing to Audio and Acoustics.
[23] Patrick A. Naylor,et al. Speech Dereverberation , 2010 .
[24] Marc Moonen,et al. Subspace Methods for Multimicrophone Speech Dereverberation , 2003, EURASIP J. Adv. Signal Process..
[25] Tomohiro Nakatani,et al. Spectral Subtraction Steered by Multi-Step Forward Linear Prediction For Single Channel Speech Dereverberation , 2006, 2006 IEEE International Conference on Acoustics Speech and Signal Processing Proceedings.
[26] Biing-Hwang Juang,et al. Speech Dereverberation Based on Probabilistic Models of Source and Room Acoustics , 2006, 2006 IEEE International Conference on Acoustics Speech and Signal Processing Proceedings.
[27] Biing-Hwang Juang,et al. Mixture autoregressive hidden Markov models for speech signals , 1985, IEEE Trans. Acoust. Speech Signal Process..
[28] Biing-Hwang Juang,et al. Blind speech dereverberation with multi-channel linear prediction based on short time fourier transform representation , 2008, 2008 IEEE International Conference on Acoustics, Speech and Signal Processing.
[29] Gary W. Elko,et al. Superdirectional microphone arrays , 2000 .
[30] Marc Delcroix,et al. On robust inverse filter design for room transfer function fluctuations , 2006, 2006 14th European Signal Processing Conference.
[31] Takuya Yoshioka,et al. Dereverberation by Using Time-Variant Nature of Speech Production System , 2007, EURASIP J. Adv. Signal Process..
[32] Tomohiro Nakatani,et al. Harmonicity-Based Blind Dereverberation for Single-Channel Speech Signals , 2007, IEEE Transactions on Audio, Speech, and Language Processing.
[33] Sadaoki Furui,et al. Digital Speech Processing, Synthesis, and Recognition , 1989 .
[34] Marc Delcroix,et al. Precise Dereverberation Using Multichannel Linear Prediction , 2007, IEEE Transactions on Audio, Speech, and Language Processing.