Signal Processing - Special issue on acoustic echo and noise control
Evaluation of an ERB frequency scale noise reduction for hearing aids: a comparative study
Speech Communication - Special issue on speech processing for hearing aids
Strategy-selective noise reduction for binaural digital hearing aids
Speech Communication - Special issue on speech processing for hearing aids
An adaptive filter-bank equalizer for speech enhancement
Signal Processing
Digital Speech Transmission: Enhancement, Coding And Error Concealment
Digital Speech Transmission: Enhancement, Coding And Error Concealment
Uniform and warped low delay filter-banks for speech enhancement
Speech Communication
Signal processing in high-end hearing aids: state of the art, challenges, and future trends
EURASIP Journal on Applied Signal Processing
Multichannel dynamic-range compression using digital frequency warping
EURASIP Journal on Applied Signal Processing
Speech and Audio Processing in Adverse Environments
Speech and Audio Processing in Adverse Environments
Speech Enhancement
Model-based feature enhancement for reverberant speech recognition
IEEE Transactions on Audio, Speech, and Language Processing - Special issue on processing reverberant speech: methodologies and applications
Model-based dereverberation preserving binaural cues
IEEE Transactions on Audio, Speech, and Language Processing - Special issue on processing reverberant speech: methodologies and applications
Hi-index | 0.00 |
A new system for single-channel speech enhancement is proposed which achieves a joint suppression of late reverberant speech and background noise with a low signal delay and low computational complexity. It is based on a generalized spectral subtraction rule which depends on the variances of the late reverberant speech and background noise. The calculation of the spectral variances of the late reverberant speech requires an estimate of the reverberation time (RT) which is accomplished by a maximum likelihood (ML) approach. The enhancement with this blind RT estimation achieves almost the same speech quality as by using the actual RT. In comparison to commonly used post-filters in hearing aids which only perform a noise reduction, a significantly better objective and subjective speech quality is achieved. The proposed system performs time-domain filtering with coefficients adapted in the non-uniform (Bark-scaled) frequency-domain. This allows to achieve a high speech quality with low signal delay which is important for speech enhancement in hearing aids or related applications such as hands-free communication systems.