Data processing: speech signal processing – linguistics – language – Speech signal processing – For storage or transmission
Reexamination Certificate
2000-11-13
2004-01-20
Dorvil, Richemond (Department: 2654)
Data processing: speech signal processing, linguistics, language
Speech signal processing
For storage or transmission
C704S220000, C704S263000, C704S228000
Reexamination Certificate
active
06681202
ABSTRACT:
FIELD OF THE INVENTION
The invention relates to digital transmission systems and more particularly to a system for enabling at the receiving end to extend a speech signal received in a narrow band, for example the telephony band (300-3400 Hz) into an extended speech signal in a wider band (for example 100-7000 Hz).
BACKGROUND ART
Most current telecommunication systems transmit a speech bandwidth limited to 300-3400 Hz (narrow band speech). This is sufficient for a telephone conversation but natural speech bandwidth is much wider (100-7000 Hz). Actually, the low band (100-300 Hz) and the high band (3400-7000 Hz) are important for listening comfort, speech naturalness and for better recognizing the speaker voice. The regeneration of these frequency bands at a phone receiver would thus enable to strongly improve speech quality in telecommunication systems. Moreover, during a phone conversation, speech is often corrupted by background noise especially when mobile phones are used. Also, the telephone network may transmit music played by switchboards. Therefore, the system that generates the low band and high band should both fit as much as possible to speech and should allow to reduce noise and improve music subjective quality.
The U.S. Pat. No. 5,581,652 describes a Code book Mapping method for extending the spectral envelope of a speech signal towards low frequencies. According to this method, low band synthesis filter coefficients are generated from narrow band analysis filter coefficients thanks to a training procedure using vector quantization as described in the article by Y. Linde, A. Buzo, R. M. Gray: “An algorithm for Vector Quantizer Design”, IEEE Transactions on Communications, Vol. COM-28, No 1, January 1980. The training procedure allows to compute two different code books: an extended one for the extended frequency band and a narrow one for the narrow band. Said narrow code book is computed from the extended code book using vector quantization so that each vector of the extended code book is linked with a vector of the narrow band code book. Then the coefficients of the low band synthesis filter are computed from these code books.
However, this method presents some drawbacks, which are responsible for the production of a rattling background sound. First the number of synthesis filter shapes is limited to the size of the code books. Second the extracted vectors in the extended band are not very correlated with the vectors obtained from the linear prediction of the narrow band speech signal. Another method called extension matrix was thus developed in order to improve signal quality at the receiving end.
SUMMARY OF THE INVENTION
It is an object of the invention to provide a method for extending at the receiving end a narrow band speech signal into a wider band speech signal in order to increase signal naturalness and listening comfort which yields to a better signal quality. The invention is particularly advantageous in telephony systems.
In accordance with the invention, the received speech signal is detected with respect to a specific speech characteristic before an extension matrix is applied to the signal, said extension matrix having coefficients depending on said detected characteristic.
In a preferred embodiment of the invention, said specific characteristic called voicing relates to the detected presence of voiced/unvoiced sounds in the received speech signal which can be detected by known methods such as the one described in the manual “Speech Coding and Synthesis”, by W. B. Kleijn and K. K. Paliwal, published by Elsevier in 1995. Then the matrixes are computed from a data base, said data base being split with respect to the detected voicing, by applying an algorithm based on Least Squared Error criterion on Linear Prediction Coding (LPC) parameters as described by C. L. Lawson and R. J. Hanson, in “Solving Least Squares Problems”, Prentice-Hall, 1974, or based on the Constrained Least Square method described in “Practical Optimization” by P. E. Gill, W. Murray and M. H. Wright published by Academic Press, London 1981.
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C.L. Lawson et al., “Solving Least Squares Problems”, Prentice Hall Jun. 1974.
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Epps J, et al. : Entitled: “A New Technique for Wideband Enhancement of Coded Narrowband Speech” IEE Workshop on Speech Coding. Model, Coders, and Error Criteria, Porvoo, Finland, Jun. 20-23, 1999, pp. 174-176.
By Miet G, et al.: Entitled: “Low-Band Extension of Telephone-Band Speech”, IEEE International Conference on Acoustics, Speech, and Signal Processing, Istanbul, Turkey, Jun. 5-9, 2000, pp. 1851-1854 vol. 3.
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Gerrits Andy
Miet Giles
Azad Abul K.
Dorvil Richemond
Koninklijke Philips Electronics , N.V.
Slobod Jack D.
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