Data processing: speech signal processing – linguistics – language – Speech signal processing – For storage or transmission
Reexamination Certificate
2000-09-06
2003-05-13
To, Doris H. (Department: 2655)
Data processing: speech signal processing, linguistics, language
Speech signal processing
For storage or transmission
C704S228000, C704S222000
Reexamination Certificate
active
06564184
ABSTRACT:
TECHNICAL FIELD
The present invention relates to a digital filter design, and especially to filter design in the frequency domain.
BACKGROUND
There are several applications in which digital filters H[k] are designed “on the fly” in the frequency domain. One example is noise suppression using spectral subtraction. Another example is design of frequency selective non-linear processors for echo cancellation. A characteristic feature of such applications is that the filter design method is quite complex. Since the is filters are updated frequently this puts a heavy burden on the hardware/software that implements these design algorithms.
Reference [1] describes a method that divides the frequency domain k into segments of equal or unequal length, and uses a constant value in each segment for H[k] and the underlying power spectral density estimates &PHgr;
x
[k] of the noisy or echo contaminated speech signal. This reduces the complexity, since the filter H[k] only has to be determined for the frequency segments and not for each frequency bin k. However, this method also has the draw-back that it may split a peak of H[k] into two different segments. This may lead to fluctuating peaks, which produces annoying “music noise”. It also reduces spectral sharpness, which further reduces speech quality.
SUMMARY
An object of the present invention is to reduce or eliminate these drawbacks of the prior art.
This object is achieved in accordance with the attached claims.
Briefly, the present invention dynamically adapts the segment lengths and positions to the current shape of the power spectrum of the speech signal. The peaks and valleys of the spectrum are determined, and the method makes sure that peaks are not split between different segments when the segments are distributed over the frequency domain. Preferably each peak is covered by a segment centered on the peak. The segment length is preferably controlled by the frequency characteristics of the human auditory system.
This method has the advantage of reducing the complexity of the filter calculation without sacrificing accuracy at the important spectrum peaks. Furthermore, the method also reduces the variance of the spectrum from frame to frame, which improves speech quality.
REFERENCES:
patent: 4658426 (1987-04-01), Chabries et al.
patent: 4811398 (1989-03-01), Copperi et al.
patent: 5394473 (1995-02-01), Davidson
patent: 5729694 (1998-03-01), Holzrichter et al.
patent: 5839101 (1998-11-01), Vähätalo et al.
patent: 5864806 (1999-01-01), Mokbel et al.
patent: 96/24128 (1996-08-01), None
Van de Wouw et al (“Statistical Bilinearization In Stochastic Nonlinear Dynamics”, 2nd International Conference on Control of Oscillations and Chaos, Jul. 2000).*
Chuang et al (“Adaptive Two-Band Spectral Subtraction With Multi-Window Spectral Estimation”, International Conference on Acoustics, Speech, and Signal Processing, Mar. 1999).*
Flores et al (“Continuous Speech Recognition In Noise Using Spectral Subtraction And HMM Adaptation”, IEEE International Conference on Acoustics, Speech, and Signal Processing, Apr. 1994).*
Sheikhzadeh et al (“Comparative Performance Of Spectral Subtraction And HMM-Based Speech Enhancement Strategies with Application to Hearing And Design”, IEEE International Conference on Acoustics, Speech, and Signal Processing, Apr. 1994).*
Matsumoto et al (“Smoothed Spectral Subtraction For A Frequency-Weighted HMM In Noisy Speech Recognition”, ICSLP Fourth International Conference on Spoken Language, Oct. 1996).*
S.F. Boll, “Suppression of acoustic noise in speech using spectral subtraction”, IEEE Trans. on Acoustics, Speech and Signal Processing, vol. ASSP-27, No. 2, Apr. 1979, pp. 113-120.
J.S. Lim and A.V. Oppenheim, “Enhancement and bandwidth compression of noisy speech”, Proc. of IEEE, vol. 67, No. 12, Dec. 1979, pp. 1586-1604.
U. Zölser, “Digital audio signal processing”, John Wiley & Sons, Chichester, U.K., 1997 pp. 252-253.
Nolan Daniel
Telefonaktiebolaget LM Ericsson (publ)
To Doris H.
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