Adaptive signal processing
Adaptive filter theory (3rd ed.)
Adaptive filter theory (3rd ed.)
ICASSP '95 Proceedings of the Acoustics, Speech, and Signal Processing, 1995. on International Conference - Volume 02
Speech enhancement based on a priori signal to noise estimation
ICASSP '96 Proceedings of the Acoustics, Speech, and Signal Processing, 1996. on Conference Proceedings., 1996 IEEE International Conference - Volume 02
Subband affine projection algorithm for acoustic echo cancellation system
EURASIP Journal on Applied Signal Processing
An embedding approach to frequency-domain and subband adaptivefiltering
IEEE Transactions on Signal Processing
IEEE Transactions on Signal Processing
Adaptive filtering in subbands using a weighted criterion
IEEE Transactions on Signal Processing
Fast Newton transversal filters-a new class of adaptive estimationalgorithms
IEEE Transactions on Signal Processing
Alias-Free Subband Adaptive Filtering With Critical Sampling
IEEE Transactions on Signal Processing
IEEE Transactions on Signal Processing
A new approach to subband adaptive filtering
IEEE Transactions on Signal Processing
Blind source separation based on a fast-convergence algorithm combining ICA and beamforming
IEEE Transactions on Audio, Speech, and Language Processing
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This paper addresses the problem of acoustic noise reduction and speech enhancement by adaptive filtering algorithms. Most speech enhancement methods and algorithms which use adaptive filtering structure are generally expressed in fullband form. One of these widespread structures is the Forward Blind Source Separation Structure (FBSS). This FBSS structure is often used to separate speech form noise and therefore enhance the speech signal at the processing output. In this paper, we propose a new subband implementation of this FBSS structure. In order to give more robustness to the proposed structure, we adapt then we apply to this subband structure a new combination of criteria based on the system mismatch and the smoothing filtering errors minimizations. The combination between this proposed subband structure with this optimal criteria allows to obtain a new two-channel subband forward (2CSF) algorithm that improves the convergence speed of the cross adaptive filters which are used to separate speech from noise. Objective tests under various environments are presented showing the good behavior of the proposed 2CSF algorithm.