Independent thesis Advanced level (degree of Master (Two Years)), 20 credits / 30 HE credits
Speech enhancement systems achieving a joint suppression of reverberation and background
noise can be used in digital hearing aids, voice controlled systems or hands-free
telephony. Demanding requirements for computational complexity, signal delay and speech
quality must be fulfilled in order to achieve a satisfactory performance. The speech quality
depends on how accurate the reverberation characteristics such as the reverberation time
or the spectral variance of the late reverberant speech are estimated. In this thesis, an
efficient algorithm for a blind reverberation time estimation based on maximum likelihood
approach is introduced. The new algorithm allows to estimate reverberation times from
a much wider range with acceptable accuracy. Variance of the late reverberant speech
is another important quantity in dereverberation systems. Two late reverberant spectral
variance estimation methods are compared with regard to estimation accuracy and computational
complexity. Finally, the performance of the considered speech enhancement
system is analyzed with the improved reverberation time estimator.
2010. , 61 p.