Speech dereverberation via maximum-kurtosis subband adaptive filtering

ICASSP, IEEE International Conference on Acoustics, Speech and Signal Processing - Proceedings

Volumn 6, Issue , 2001, Pages 3701-3704

  Gillespie, Bradford W ^a   Malvar, Henrique S ^b   Florêncio, Dinei A F ^b  

^a University of Washington   (United States)

^b MICROSOFT RESEARCH   (United States)

Author keywords

[No Author keywords available]

Indexed keywords

ACOUSTIC SIGNAL PROCESSING; ADAPTIVE ALGORITHMS; CONVERGENCE OF NUMERICAL METHODS; MATHEMATICAL MODELS; MATHEMATICAL TRANSFORMATIONS; MICROPHONES; SIGNAL RECONSTRUCTION; SPEECH ANALYSIS; SPEECH SYNTHESIS; TIME DOMAIN ANALYSIS;

KURTOSIS; LINEAR PREDICTION; MODULATED COMPLEX LAPPED TRANSFORM; SPEECH DEREVERBERATION; SUBBAND ADAPTIVE FILTERING;

ADAPTIVE FILTERING;

EID: 0034857681     PISSN: 15206149     EISSN: None     Source Type: Journal    
DOI: 10.1109/ICASSP.2001.940646     Document Type: Article

References (11)

1. Adaptive noise reduction of speech signals
2. Computer-steered microphone arrays for sound transduction in large rooms
3. An approach to dereverberation using multi-microphone sub-band envelope estimation
4. Speech dereverberation and noise reduction with a combined microphone array approach
5. On the use of explicit speech modeling in microphone array applications
6. Enhancement of reverberant speech using LP residual signal
7. Minimum entropy deconvolution
8. Performance of some sparseness criterion blind deconvolution methods in the presence of noise
9. Least-mean kurtosis: A novel higher-order statistics based adaptive filtering algorithm
10. A modulated complex lapped transform and its application to audio processing