Adaptive signal processing
Multirate systems and filter banks
Multirate systems and filter banks
Adaptive filter theory (3rd ed.)
Adaptive filter theory (3rd ed.)
Acoustic signal processing for telecommunication
Acoustic signal processing for telecommunication
ICASSP '97 Proceedings of the 1997 IEEE International Conference on Acoustics, Speech, and Signal Processing (ICASSP '97) -Volume 1 - Volume 1
Affine projection algorithm for oversampled subband adaptive filters
ICME '03 Proceedings of the 2003 International Conference on Multimedia and Expo - Volume 2
New structures for adaptive filtering in subbands with critical sampling
IEEE Transactions on Signal Processing
Convergence behavior of affine projection algorithms
IEEE Transactions on Signal Processing
A family of adaptive filter algorithms with decorrelatingproperties
IEEE Transactions on Signal Processing
Convergence analysis of alias-free subband adaptive filters based on a frequency domain technique
IEEE Transactions on Signal Processing
Mean-square performance of a family of affine projection algorithms
IEEE Transactions on Signal Processing
A new approach to subband adaptive filtering
IEEE Transactions on Signal Processing
Computers and Electrical Engineering
Computers and Electrical Engineering
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We present a new subband affine projection (SAP) algorithm for the adaptive acoustic echo cancellation with long echo path delay. Generally, the acoustic echo canceller suffers from the long echo path and large computational complexity. To solve this problem, the proposed algorithm combines merits of the affine projection (AP) algorithm and the subband filtering. Convergence speed of the proposed algorithm is improved by the signal-decorrelating property of the orthogonal subband filtering and the weight updating with the prewhitened input signal of the AP algorithm. Moreover, in the proposed algorithms, as applying the polyphase decomposition, the noble identity, and the critical decimation to subband the adaptive filter, the sufficiently decomposed SAP updates the weights of adaptive subfilters without a matrix inversion. Therefore, computational complexity of the proposed method is considerably reduced. In the SAP, the derived weight updating formula for the subband adaptive filter has a simple form as ever compared with the normalized least-mean-square (NLMS) algorithm. The efficiency of the proposed algorithm for the colored signal and speech signal was evaluated experimentally.