IEEE SIGNAL PROCESSING LETTERS, VOL. 6, NO.

3, MARCH 1999

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A Single Microphone Kalman Filter-Based Noise Canceller

M. Gabrea, E. Grivel, and M. Najim, Fellow, IEEE
Abstract A great deal of attention has been paid to speech enhancement using a single microphone system. The various approaches, based on the Kalman lter, operate in two steps: 1) the noise variances and the parameters of the speech model are estimated, and 2) the speech signal is retrieved using standard Kalman ltering. This letter presents an alternative solution that does not require the explicit estimation of the noise and the driving process variances. This deals with a new formulation of the approach proposed within a control literature framework by Mehra. Index TermsInnovation, Kalman lter, speech enhancement.

the Kalman lter. Sections III and IV, respectively, describe the speech parameter estimation and the Kalman gain calculation. The last section presents experimental results and evaluates the performance of the proposed approach. II. NOISY SPEECH MODEL AND KALMAN FILTERING The speech signal process is modeled as a th-order AR

I. INTRODUCTION IVEN a sequence of speech signal corrupted by an additive white noise, our purpose is to retrieve the speech signal. Several approaches have been reported in the literature. They essentially differ in the way they estimate both the speech model parameters and the noise variances. Kalman ltering for speech enhancement was proposed in [1], where speech parameters were obtained from the clean speech signal, before being contaminated by the noise. The method developed in [2] provides a suboptimal solution, which is a simplied version of the estimate-maximize (EM) algorithm based on the maximum likelihood argument. In [3] an adaptive algorithm which estimates the model parameters in a sequential way is proposed. The speech signal in [4] is modeled as an autoregressive moving average (ARMA) process, which is then estimated using a standard Kalman lter. In the methods mentioned above, the use of Kalman ltering always implies the estimation of the variances of both additive white noise and driving process. In this paper, we propose a Kalman lter based-method that avoids the explicit estimation of noise variances. For such a purpose, we reformulate the approach developed by Mehra in the eld of identication [5], [6]. Unlike existing methods, no voice activity detector (VAD) is required to estimate noise variance. Furthermore, Mehras algorithm operates globally, in comparison to the multistage procedures of other approaches. This work is organized as follows. Section II presents the state space model that represents the noisy speech signal and
Manuscript received November 30, 1998. The associate editor coordinating the review of this manuscript and approving it for publication was Prof. K. K. Paliwal. The authors are with Equipe Signal et Image, ENSER Bordeaux, 33402 Talence Cedex, France (e-mail: najim@goelette.tsi.u-bordeaux.fr). Publisher Item Identier S 1070-9908(99)01788-5.

(1) is the th sample of the speech signal, is the th is the th AR parameter. sample of the observation, and Equation (1) can be represented by the state space model

(2) where and are uncorrelated Gaussian 1) the sequences white noise sequences with zero means and covariances and 2) is the state vector (3) 3) A is the transition matrix . . . . . . .. .. .. . .. . . . (4)

. .

4)

and the

are, respectively, the input vector and observation row vector dened as follows: (5)

The standard Kalman lter provides the updating state vector estimator [7]. However, the transition matrix and the are unknown and hence must be estimated. Kalman gain

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IEEE SIGNAL PROCESSING LETTERS, VOL. 6, NO. 3, MARCH 1999

III. PARAMETER ESTIMATION In our approach, getting requires the AR parameter estimation. Such an issue is not addressed in this paper. This is the reason why we propose to estimate the AR parameters from the modied YuleWalker equations [8], even if they may sometimes lead to unsatisfactory performances, especially for wideband signals [9]:
SNR GAIN

FOR

TABLE I VARIOUS INPUT SNR

. . .

..

. . .

. . .

(6)

is the autocorrelation matrix of the prediction error, is the autocorrelation of the innovation obtained with , and is the identity matrix. If we pay attention to the formula (8) and compare it to as the formula (7), we can dene the quantity . If we iterate (8), one obtains

where

denotes the observation where denotes the expectation and autocorrelation function, denotes the pseudoinverse operator. However, the observation autocorrelation function must be estimated from the observed data. IV. KALMAN GAIN ESTIMATION When the standard Kalman lter gain is recursively comand Mehra puted, one needs the a priori knowledge of proposed an iterative nonoptimal computational Kalman gain in the case where these quantities are not known. By checking the whiteness of the innovation sequence one can test whether the asymptotically optimal solution has been reached or not [10]. Nevertheless one problem remains regarding the initialization of the iterative algorithm. A. Iterative Estimation of Kalman Gain In the following, we propose an alternative method for the Kalman gain iterative algorithm. Let us consider the for a timecalculation of the optimal Kalman gain invariant optimal lter. It can be shown that [11] (7) where is the prediction error autocorrelation matrix and is the autocorrelation of the innovation obtained Let be the initial value of the Kalman gain. with The following relationship is easily obtained: . . .

. . .

(9)

Equation (9) is identical to the one directly proposed by Mehra [5], [6]. The convergence of (9) has been demonstrated in [5]. B. Statistical Test The state vector estimate and the Kalman gain are updated until the innovation is a white process. Once the innovation is white, the estimated speech signal can be retrieved from the equation. (10) V. SIMULATIONS
AND

RESULTS

. . .

. . .

(8)

The efciency of the method is tested on a natural speech signal corrupted by a Gaussian white noise for various input signal-to-noise ratio (SNR). Table I allows the comparison with classical approaches, by giving averaged SNR gain based on 100 realizations, for various SNR gain from 5 to 15 dB. The approach developed in [1] cannot lead to practical implementation, but stands as an ideal case in our simulations. For input SNR between 5 and 15 dB our method, that converges rapidly (two to four iterations), provides better results than Oppenheims one. Gibsons algorithm, which needs two to three iterations to get the highest SNR gain, allows to obtain SNR gains that are always higher than ours; however, its computational requirement is higher, since a VAD is required to determine silence periods. In addition, our algorithm has the advantage to operate globally, in comparison with the multistage procedures of others approaches.

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REFERENCES
[1] K. K. Paliwal and A. Basu, A speech enhancement method based on Kalman ltering, in Proc. ICASSP87, pp. 177180. [2] J. D. Gibson, B. Koo, and S. D. Gray, Filtering of colored noise for speech enhancement and coding, IEEE Trans. Signal Processing, vol. 39, pp. 17321742, Aug. 1991. [3] A. V. Oppenheim et al., Single-sensor active noise cancellation, IEEE Trans. Speech Audio Processing, vol. 2, pp. 285290, Apr. 1994. [4] H. Morikawa and H. Fujisaki, Noise reduction of speech signal by adaptive Kalman ltering, APII-AFCET, Special Issue Signal Process., vol. 22, pp. 5368, 1988. [5] R. K. Mehra, On the identication of variances and adaptive Kalman ltering, IEEE Trans. Automat. Contr., vol. AC-15, pp. 175184, Apr. 1970.

[6] [7] [8] [9] [10] [11]

, On-line identication of linear dynamic systems with applications to Kalman ltering, IEEE Trans. Automat. Contr., vol. AC-16, pp. 1221, Feb. 1971. M. Najim, Modelization and Identications in Signal Processing. Paris, France: Masson, 1988 (in French). C. W. Therrien, Discrete Random Signals and Statistical Signal Processing. Englewood Cliffs, NJ: Prentice-Hall, 1992. K. K. Paliwal, A noise-compensated long correlation matching method for AR spectral estimation of noisy signals, in Proc. IEEE ICASSP86, pp. 13691372. T. Kailath, An innovations approach to least-squares estimation, part I: Linear ltering in additive white noise, IEEE Trans. Automat. Contr., vol. AC-13, pp. 646655, Dec. 1968. B. A. Anderson and J. B. Moore, Optimal Filtering. Englewood Cliffs, NJ: Prentice-Hall 1979.