Strategy-selective noise reduction for binaural digital hearing aids
Speech Communication - Special issue on speech processing for hearing aids
Digital Speech Transmission: Enhancement, Coding And Error Concealment
Digital Speech Transmission: Enhancement, Coding And Error Concealment
Signal processing in high-end hearing aids: state of the art, challenges, and future trends
EURASIP Journal on Applied Signal Processing
Time delay estimation in room acoustic environments: an overview
EURASIP Journal on Applied Signal Processing
Dual-channel speech enhancement by superdirective beamforming
EURASIP Journal on Applied Signal Processing
Enhancement of reverberant speech using the CELP postfilter
ICASSP '09 Proceedings of the 2009 IEEE International Conference on Acoustics, Speech and Signal Processing
Binaural artificial bandwidth extension (B-ABE) for speech
ICASSP '09 Proceedings of the 2009 IEEE International Conference on Acoustics, Speech and Signal Processing
Efficient musical noise suppression for speech enhancement system
ICASSP '09 Proceedings of the 2009 IEEE International Conference on Acoustics, Speech and 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
A binaural room impulse response database for the evaluation of dereverberation algorithms
DSP'09 Proceedings of the 16th international conference on Digital Signal Processing
A two-stage algorithm for one-microphone reverberant speech enhancement
IEEE Transactions on Audio, Speech, and Language Processing
Generalized Spherical Array Beamforming for Binaural Speech Reproduction
IEEE/ACM Transactions on Audio, Speech and Language Processing (TASLP)
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The ability of the human auditory system for sound localization mainly depends on the binaural cues, especially interaural time and level differences (ITD and ILD). In the context of digital hearing aids and binaural audio transmission systems, these cues can be severely degraded by independent bilateral signal processing such as dereverberation or noise reduction. This contribution presents a novel two-stage binaural dereverberation algorithm which explicitly preserves the binaural cues. The first stage is based on a statistical model of the room impulse responses (RIR) and comprises a spectral subtraction rule which reduces late reverberation only. It includes a smoothing process of the spectral gains to reduce musical tones. In a second stage, the residual reverberation is attenuated by a dual-channel Wiener filter. This is derived from a coherence model of the reverberant sound field taking into account shadowing effects of the head. The overall binaural-input binaural-output structure efficiently reduces both early and late reverberation. In experiments as well as informal listening tests using measured binaural room impulse responses, the proposed algorithm significantly improves speech quality according to objective and subjective measures.