Multirate systems and filter banks
Multirate systems and filter banks
Digital signal processing (3rd ed.): principles, algorithms, and applications
Digital signal processing (3rd ed.): principles, algorithms, and applications
Signal Processing - Special issue on acoustic echo and noise control
Discrete-time signal processing (2nd ed.)
Discrete-time signal processing (2nd ed.)
Multirate Digital Signal Processing
Multirate Digital Signal Processing
An adaptive filter-bank equalizer for speech enhancement
Signal Processing
Psychoacoustics: Facts and Models
Psychoacoustics: Facts and Models
ICASSP '99 Proceedings of the Acoustics, Speech, and Signal Processing, 1999. on 1999 IEEE International Conference - Volume 02
Speech enhancement by map spectral amplitude estimation using a super-Gaussian speech model
EURASIP Journal on Applied Signal Processing
Multichannel dynamic-range compression using digital frequency warping
EURASIP Journal on Applied Signal Processing
IEEE Transactions on Signal Processing
Low delay noise reduction and dereverberation for hearing aids
EURASIP Journal on Advances in Signal Processing - Special issue on digital signal processing for hearing instruments
Noise suppression using one-regular unimodular filterbanks
ICME'09 Proceedings of the 2009 IEEE international conference on Multimedia and Expo
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A versatile filter-bank concept for adaptive subband filtering is proposed, which achieves a significantly lower algorithmic signal delay than commonly used analysis-synthesis filter-banks. It is derived as an efficient implementation of the filter-bank summation method and performs time-domain filtering with coefficients adapted in the uniform or non-uniform frequency-domain. The frequency warped version of the proposed filter-bank has a lower computational complexity than the usual warped analysis-synthesis filter-bank for most parameter configurations. The application to speech enhancement shows that the same quality of the enhanced speech can be achieved but with lower signal delay. For systems with tight signal delay requirements, modifications of the new filter-bank design are discussed to further decrease its signal delay by approximating the original time-domain filter by an FIR or IIR filter of lower degree. This approach can achieve a very low signal delay and reduced computational complexity with almost no loss for the perceived speech quality.