Adaptive post-filtering technique based on the Modified...

Details Adaptive post-filtering technique based on the Modified... Adaptive post-filtering technique based on the Modified...

1999-03-12

2001-05-15

Dorvil, Richemond (Department: 2741)

Data processing: speech signal processing, linguistics, language

Speech signal processing

For storage or transmission

C704S219000

Reexamination Certificate

active

06233552

ABSTRACT:

BACKGROUND OF THE INVENTION
A perfect post-filtering technique should not alter the formant information and should attenuate null information in the speech spectrum in order to achieve noise reduction and hence produce better speech quality. Conventionally, time-domain post-filtering techniques use modified LPC synthesis, inverse, and high pass filters that are derived from an LPC spectrum and are configured by the constants: &agr; (for modified synthesis filter), &bgr; (for modified inverse filter) and &mgr; (for high pass filter). See, Juiun-Hwey Chen, Allen Gersho “Adaptive Post-filtering For Quality Enhancement of Coded Speech”, IEEE Trans. Speech & Audio Proc., vol. 3, no. 1, pp. 59-71, 1995. Such a filter has been used successfully in low bit rate coders, but it is very hard to adapt the coefficients from one frame to another and still produce a post-filter frequency response without spectral tilt. The result is time-domain post-filtering which produces varying and unpredictable spectral tilt from one frame to another which causes unnecessary attenuation or amplification of some frequency components, and a muffling of speech quality. This effect increases when voice coders are tandemed together. However, it is very hard to adapt these coefficients from one frame to another and still produce a post-filter frequency response without spectral tilt. Conventional time-domain post-filtering produces varying spectral tilt from one frame to another affecting speech quality.
Another problem with conventional time-domain post-filtering is that, when two formants are close together, the frequency response may have a peak rather than a null between the two formants hence altering the formant information. Yet another effect is that in the original speech, the first formant may have a much higher peak than the second formant, however, the frequency response of the post-filter may have a second formant with a higher peak than the first formant. These phenomena are completely undesirable because they affect the output speech quality.
Another approach of designing a post-filter is described by R. McAulay, T. Parks, T. Quatieri, M. Sabin “Sine-Wave Amplitude Coding At Low Data Rates”, Advances in Speech Coding, Kluwer Academic Pub., 1991, edited by B. S. Atal, V. Cuperman and A. Gersho, pp. 203-214. This technique has produced good performance without spectral tilt, but it can only be used in sinusoidal based speech coders.
SUMMARY OF THE INVENTION
It is, therefore, an object of the invention to provide a new time-domain post-filtering technique which eliminates the problems above, particularly the problem of spectral tilt in speech spectrum, and that can be applied to various speech coders, including both time and frequency domain speech coders.
This and other objects are achieved according to the present invention by a post-filter design approach which uses the pole information in the LPC spectrum and finds the relation between poles and formants.
The locations of poles of an LPC spectrum of said speech signal are determined, the location and bandwidth of formants of said speech signal are estimated based on the pole information, by first arranging the poles in a predetermined order (e.g., according to increasing radius) and applying an estimation algorithm to the ordered poles. The filter coefficients are estimated, a desired filter response characteristic is compared to the filter response characteristic resulting from said estimated filter coefficients to obtain a difference value, the filter coefficients are adjusted to minimize said difference value according to a least squares approach.
In accordance with a preferred embodiment of the invention, the formant estimation algorithm comprises calculating a magnitude and slope of said LPC spectrum at at least some of said arranged poles, calculating first and second slopes m1 and m2, respectively, of said LPC spectrum on either side of the arranged poles, and then (i) estimating first and second adjacent poles to represent different formants if m1 is less than zero and if m2 is greater than zero, (ii) estimating first and second adjacent poles to represent a common formant if the criteria of step (i) are not met and if a difference in magnitudes of said LPC spectrum is less than a threshold value, e.g., 3 dB, and (iii) estimating the larger of said first and second poles to represent a formant if the criteria of steps (i) and (ii) are not met. If the bandwidths assigned to adjacent formants in this process are overlapping, the formants are combined into a single bandwidth.
In accordance with the present invention, the filter is a Modified Yule-Walker (MYW) filter with a filter response given by:
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A
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b
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N
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a
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a
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where N is the order of the MYW filter. The (MYW) filter coefficients are estimated using a least squares fit in the time domain. The denominator coefficients of the filter (a(1), a(2), . . . , a(N)) are computed by the Modified Yule-Walker equations using non-recursive correlation coefficients computed by inverse Fourier transformation of the specified frequency response of the post-filter. The numerator coefficients of the filter (b(1), b(2), . . . , b(N)) are computed by a 4 step procedure: first, a numerator polynomial corresponding to an additive decomposition of the power frequency response is computed. The complete frequency response corresponding to the numerator and denominator polynomials is then evaluated. As a result, a spectral factorization technique is used to obtain the impulse response of the filter. Finally, the numerator polynomial is obtained by a least squares fit to this impulse response.
Test results show that the post-filter according to the present invention outperforms the conventional post-filter in both 1 and 2 tandem connection cases of the voice coders.


REFERENCES:
patent: 4764963 (1988-08-01), Atal
patent: 4945568 (1990-07-01), Willems
patent: 5054085 (1991-10-01), Meisel et al.
patent: 5235669 (1993-08-01), Ordentlich et al.
patent: 5615298 (1997-03-01), Chen
patent: 5649054 (1997-07-01), Oomen et al.
patent: 5675701 (1997-10-01), Kleijn et al.
patent: 5706394 (1998-01-01), Wynn
patent: 5708754 (1998-01-01), Wynn
patent: 5729694 (1998-03-01), Holzrichter et al.
patent: 5778338 (1998-07-01), Jacobs et al.
patent: 5781883 (1998-07-01), Wynn
patent: 5884010 (1999-03-01), Chen et al.
patent: 6026357 (2000-02-01), Ireton et al.
patent: 6041297 (2000-03-01), Goldberg

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