Adaptive Algorithms for Enhancement of Speech Subject to a High-Level Noise

• Autorzy: Latos M. , Pawełczyk M.
• Języki publikacji: EN

Abstrakty

EN

There are many industrial environments which are exposed to a high-level noise, sometimes much higher than the level of speech. Verbal communication is then practically unfeasible. In order to increase the speech intelligibility, appropriate speech enhancement algorithms can be used. It is impossible to filter off the noise completely from the acquired signal by using a conventional filter, because of two reasons. First, the speech and the noise frequency contents are overlapping. Second, the noise properties are subject to change. The adaptive realisation of the Wienerbased approach can be, however, applied. Two structures are possible. One is the line enhancer, where the predictive realisation of the Wiener approach is used. The benefit of using this structure it that it does not require additional apparatus. The second structure takes advantage of the high level of noise. Under such condition, placing another microphone, even close to the primary one, can provide a reference signal well correlated with the noise disturbing the speech and lacking the information about the speech. Then, the classical Wiener filter can be used, to produce an estimate of the noise based on the reference signal. That noise estimate can be then subtracted from the disturbed speech. Both algorithms are verified, based on the data obtained from the real industrial environment. For laboratory experiments the G.R.A.S. artificial head and two microphones, one at back side of an earplug and another at the mouth are used.

Słowa kluczowe

EN

speech enhancement; adaptive system; line enhancer; LMS algorithm; high-level noise; nonstationary noise; earplug; active noise control

• Wydawca: Instytut Podstawowych Problemów Techniki PAN ; Komitet Akustyki PAN ; Polskie Towarzystwo Akustyczne
• Czasopismo: Archives of Acoustics
• Rocznik: 2010
• Tom: Vol. 35, No. 2
• Strony: 203--212
• Opis fizyczny: Bibliogr. 8 poz., wykr.

Twórcy

autor

Latos M.

autor

Pawełczyk M.

• Silesian University of Technology Institute of Automatic Control Akademicka 16, 44-100 Gliwice, Poland,

Bibliografia

• 1. Benesty J., Sondhi M.M., Huang Y., [Eds.] (2008), Springer Handbook of Speech Processing, Springer-Verlag, Berlin Heilderberg.
• 2. Elliott S. (2001), Signal Processing for Active Control, Academic Press, London.
• 3. Haykin S. (1986), Adaptive Filter Theory, Prentice Hall, New Jersey.
• 4. Michalczyk M.I. (2004), Adaptive Control Algorithms for Three-Dimensional Zones of Quiet, Jacek Skalmierski Computer Studio, Gliwice.
• 5. Pawełczyk M. (2005), Feedback Control of Acoustic Noise at Desired Locations, Silesian University of Technology, Gliwice.
• 6. Saxena G., Ganesan S., Das M. (2008), Real-time Implementation of Adaptive Noise Cancelation, IEEE International Conference on Electro/Information Technology, 431-436.
• 7. Widrow B., Glover J.R., McCool J.M., Kaunitz J., Williams C.S., Hearn R.H., Zeidler J.R., Dong E., Goodlin R.C. (1975), Adaptive Noise Cancelling: Principles and Applications, Proceedings of the IEEE, 63, 12, 1692-1716.
• 8. Widrow B., Stearns S.D. (1985), Adaptive Signal Processing, Prentice Hall, New Jersey.