Reverberation Phenomena in Speech Signal Enhancement

Resource Overview

Speech signals often exhibit reverberation effects during enhancement processes. To eliminate background reverberation, speakers frequently need to adjust their head orientation, causing continuous variations in impulse responses. We combine blind deconvolution and spectral subtraction methods to improve inverse filter performance. By calculating correlation coefficient matrices between input speech signals, we derive stable and accurate inverse filters of room impulse responses without requiring direct measurements. This inverse filtering eliminates early reflections containing most reverberation energy, followed by spectral subtraction suppressing residual tail reverberation. Experimental validation demonstrates the practical adaptability of this combined approach compared to individual methods.

Detailed Documentation

Under normal circumstances, speech signal enhancement often introduces reverberation effects. To eliminate background reverberation, speakers frequently need to constantly change their head orientation, leading to continuous variations in impulse responses. To improve the filtering performance of inverse filters, we integrate blind deconvolution with spectral subtraction methods. Through computation of correlation coefficient matrices between input speech signals, we obtain stable and precise inverse filters for room impulse responses without requiring actual measurement of room impulse responses. The inverse filtering process eliminates early reflections that contain the majority of reverberation energy. Subsequently, we employ spectral subtraction to suppress the tail reverberation remaining in the inverse-filtered signal. Experimental validation confirms the practical adaptability of this approach, demonstrating that the combination of blind deconvolution and spectral subtraction provides superior speech environments compared to using either method individually.

Algorithm Implementation Details: - Blind deconvolution algorithm estimates inverse filters using second-order statistics of correlated speech signals - Spectral subtraction implementation involves calculating noise floor estimates and applying frequency-domain gain functions - Correlation matrix computation employs autocorrelation and cross-correlation analysis of multiple speech inputs - Inverse filter design utilizes least-squares optimization with regularization constraints - Real-time adaptation mechanisms adjust filter parameters based on changing acoustic conditions