Abstract
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.
Highlights
Algorithms for the enhancement of acoustically disturbed speech signals have been the subject of intensive research over the last decades, cf., [1,2,3]
The room impulse response (RIR) has been measured in a highly reverberant room and possesses a reverberation time (RT) of 0.79 s. (This value for T60 has been determined from the measured RIR by a modified Schroeder method as described in [43].) The reverberant speech signal z(k) is distorted by additive babble noise from the NOISEX92 database with varying global input signal-to-noise ratio (SNR) for anechoic speech s(k) and additive noise v(k)
A new speech enhancement algorithm for the joint suppression of late reverberant speech and background noise is proposed which addresses the special requirements of hearing aids
Summary
Algorithms for the enhancement of acoustically disturbed speech signals have been the subject of intensive research over the last decades, cf., [1,2,3]. Coherence-based speech enhancement algorithms such as [14] or [15] can suppress background noise and reverberation, but they are rather ineffective if only two closely spaced microphones can be used This problem can be alleviated to some extend by a noise classification and binaural processing [16] which, requires two hearing aid devices connected by a wireless data link. A single-channel algorithm for speech dereverberation and noise reduction has been proposed recently in [17] This algorithm is less suitable for hearing aids due to its high computational complexity and signal delay as well as its strong speech distortions. A single-channel speech enhancement algorithm is proposed, which is more suitable for current hearing aid devices It performs a suppression of background noise and late reverberant speech by means of a generalized spectral subtraction.
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