DocumentCode
1693419
Title
Non-intrusive whitening of speech using least mean square and divergence detection technique
Author
Ng, Wai Pang ; Elmirghani, Jaafar M H ; Cryan, R.A. ; Broom, Simon
Author_Institution
Sch. of Eng., Univ. of Northumbria, Newcastle, UK
Volume
4
fYear
1999
Firstpage
2203
Abstract
A speech whitening technique is presented and used for improved echo path modelling in telephony networks. The system identification of interest is based on the real time least mean square (LMS) algorithm and a class of digital adaptive filters (DAFs). The modelling convergence rate derived from the optimal Wiener weights defines the performance criterion. A novel non-intrusive whitening technique based on the speech characteristics is exploited to whiten the speech power spectral density (PSD) whilst preserving the signal bandwidth requirements. The technique involves pre-filtering the speech using tap weight coefficients of the inverse speech spectrum. Software simulation shows an improved performance compared to the conventional LMS. A new divergence detection (DD) technique is used in a noise-impaired environment to eliminate divergence by controlling the adaptation process. The DD technique reported produces significant performance improvement in noisy environments and at echo to noise ratios (e/N) of up to 0 dB. The combined improvement reported using the whitening technique and DD is 24.5 dB after 8000 iterations (1 second) at e/N of 0 dB.
Keywords
Wiener filters; adaptive filters; adaptive signal detection; adaptive signal processing; convergence of numerical methods; digital filters; digital simulation; filtering theory; least mean squares methods; noise; spectral analysis; speech processing; telephone networks; LMS; PSD; digital adaptive filters; divergence detection; echo path modelling; echo to noise ratios; inverse speech spectrum; least mean square; modelling convergence rate; noise-impaired environment; noisy environments; nonintrusive speech whitening; optimal Wiener weights; performance criterion; power spectral density; real time LMS algorithm; signal bandwidth; software simulation; speech characteristics; speech pre-filtering; tap weight coefficients; telephony networks; Adaptive filters; Convergence; Least squares approximation; Power system modeling; Real time systems; Signal to noise ratio; Speech; System identification; Telephony; Working environment noise;
fLanguage
English
Publisher
ieee
Conference_Titel
Global Telecommunications Conference, 1999. GLOBECOM '99
Conference_Location
Rio de Janeireo, Brazil
Print_ISBN
0-7803-5796-5
Type
conf
DOI
10.1109/GLOCOM.1999.827595
Filename
827595
Link To Document