Two-channel post-filtering based on adaptive smoothing and noise properties

This paper studies the statistical properties of the gain functions, which are often used for two-channel post-filtering (TC-PF) algorithms. We reveal that the smoothing factor has a significant impact on both noise reduction and musical noise. When the smoothing factor increases, noise reduction can be improved and musical noise can be reduced simultaneously. However, the smoothing factor could not be too close to one because the system can only be assumed to be time-invariant for short durations. To solve this problem, this paper proposes an adaptive smoothing scheme by detecting the sudden change of the system. Moreover, the residual noise floor is adaptively chosen based on the structure of the noise power spectral density (NPSD) to further suppress the tonal noise components. Experimental results show the better performance of the proposed algorithm in terms of the segmental signal-to-noise-ratio (SNR) and the PESQ improvements.

[1]  Michael S. Brandstein,et al.  Microphone Arrays - Signal Processing Techniques and Applications , 2001, Microphone Arrays.

[2]  Donald B. Percival,et al.  Spectrum estimation by wavelet thresholding of multitaper estimators , 1998, IEEE Trans. Signal Process..

[3]  Gérard Faucon,et al.  Using the coherence function for noise reduction , 1992 .

[4]  Jesper Jensen,et al.  Adaptive Time Segmentation for Improved Speech Enhancement , 2006, IEEE Transactions on Audio, Speech, and Language Processing.

[5]  Xiaodong Li,et al.  Speech enhancement based on the structure of noise power spectral density , 2010, 2010 18th European Signal Processing Conference.

[6]  Jont B. Allen,et al.  Multimicrophone signal‐processing technique to remove room reverberation from speech signals , 1977 .

[7]  Stefan Ernst,et al.  Combination of two-channel spectral subtraction and adaptive wiener post-filtering for noise reduction and dereverberation , 1996, 1996 8th European Signal Processing Conference (EUSIPCO 1996).

[8]  Jacob Benesty,et al.  Speech Enhancement , 2010 .

[9]  Yi Hu,et al.  A comparative intelligibility study of single-microphone noise reduction algorithms. , 2007, The Journal of the Acoustical Society of America.

[10]  Rainer Martin,et al.  Bias compensation methods for minimum statistics noise power spectral density estimation , 2006, Signal Process..

[11]  Hervé Bourlard,et al.  Microphone array post-filter based on noise field coherence , 2003, IEEE Trans. Speech Audio Process..

[12]  Israel Cohen,et al.  Multichannel post-filtering in nonstationary noise environments , 2004, IEEE Transactions on Signal Processing.

[13]  Yannick Mahieux,et al.  Analysis of noise reduction and dereverberation techniques based on microphone arrays with postfiltering , 1998, IEEE Trans. Speech Audio Process..

[14]  Israel Cohen,et al.  Noise spectrum estimation in adverse environments: improved minima controlled recursive averaging , 2003, IEEE Trans. Speech Audio Process..

[15]  Chris Chatfield,et al.  The Analysis of Time Series: An Introduction , 1981 .

[16]  Petre Stoica,et al.  Total-Variance Reduction Via Thresholding: Application to Cepstral Analysis , 2007, IEEE Transactions on Signal Processing.

[17]  Rainer Martin,et al.  Noise power spectral density estimation based on optimal smoothing and minimum statistics , 2001, IEEE Trans. Speech Audio Process..

[18]  Yang Lu,et al.  A geometric approach to spectral subtraction , 2008, Speech Commun..

[19]  Parham Aarabi,et al.  Phase-Based Dual-Microphone Speech Enhancement Using A Prior Speech Model , 2007, IEEE Transactions on Audio, Speech, and Language Processing.

[20]  Jingdong Chen,et al.  Microphone Array Signal Processing , 2008 .

[21]  Ephraim.  Speech enhancement using a minimum mean square error short-time spectral amplitude estimator , 1984 .

[22]  B C Wheeler,et al.  Localization of multiple sound sources with two microphones. , 2000, The Journal of the Acoustical Society of America.

[23]  John G. Proakis,et al.  Probability, random variables and stochastic processes , 1985, IEEE Trans. Acoust. Speech Signal Process..

[24]  C. Chatfield,et al.  Fourier Analysis of Time Series: An Introduction , 1977, IEEE Transactions on Systems, Man, and Cybernetics.

[25]  R. Zelinski,et al.  A microphone array with adaptive post-filtering for noise reduction in reverberant rooms , 1988, ICASSP-88., International Conference on Acoustics, Speech, and Signal Processing.

[26]  Mohsen Rahmani,et al.  Power level difference as a criterion for speech enhancement , 2009, 2009 IEEE International Conference on Acoustics, Speech and Signal Processing.