H1 FILTERING FOR SPEECH ENHANCEMENT

Xuemin Shen, Li Deng and Anisa Yasmin Dept. of Elec. & Comp. Eng., Univ. of Waterloo, Waterloo, Ontario, Canada N2L 3G1

ABSTRACT
In this paper, a new approach based on the H1 ltering is presented for speech enhancement. This approach differs from the traditional modied Wiener/Kalman ltering approach in the following two aspects: 1) no a priori knowledge of the noise statistics is required; instead the noise signals are only assumed to have nite energy; 2) the estimation criterion for the lter design is to minimize the worst possible amplication of the estimation error signal in terms of the modeling errors and additive noises. Since most additive noises in speech are not Gaussian, this approach is highly robust and is more appropriate in practical speech enhancement. The global signal-tonoise ratio (SNR), time domain speech representation and listening evaluations are used to verify the performance of the H1 ltering algorithm. Experimented results show that the ltering performance is better than other speech enhancement approaches in the literature under similar experimental conditions.

1.

INTRODUCTION

Noise contaminated speech results in various degrees of reduction of speech discrimination. For example, background acoustic noise degrades speech signal quality of mobile telephone systems; airplane engine noise affects the conversation between a pilot and an air trafc controller. With the objective of enhancing the quality and intelligibility of speech, speech enhancement involves manipulation of the contaminated speech signal to mitigate noise effects. In order to enhance the quality and intelligibility of speech, speech enhancement involves processing of speech signal to improve these aspects. There have been numerous studies involving this subject [1]-[4]. Based on stochastic speech models, the previous studies have been focused on the minimization of the variance of the estimation error of the speech, i.e., the celebrated Wiener and/or Kalman ltering approach. This type of estimation assumes that both dynamics of signal generating processes and the statistical properties of noise sources are known in advance. However, this unrealistic assumption naturally limit the application of the estimators since in many situations only approximate signal models are available and/or the statistics of the noise sources are not fully known or unavailable. Furthermore, both Wiener and Kalman estimators may not be sufciently robust to the signal model errors. In this paper, a new approach based on the recently developed optimal ltering [5]-[10] H1 ltering is presented to speech enhancement. This approach

differs from the traditional modied Wiener/Kalman ltering approach in the following two aspects: 1) No a priori knowledge of the noise statistics is required. The only assumption is that the noise signals have nite energy; and 2) The estimation criterion in the H1 lter design is to minimize the worst possible effects of the modeling errors and additive noise on the signal estimation errors. Since the noise added to speech is not Gaussian in general, this ltering approach appears highly robust and more appropriate in practical speech enhancement. Furthermore, the H1 ltering algorithm is straightforward to implement. Our experimental results have demonstrated that the ltering performance of the H1 estimation noticeably superior to that of the Kalman estimation. The remainder of this paper is organized as follows. In Section 2, the speech source model (vocal tract) is characterized by an all pole lter. This speech source model and the observation model are then combined into a canonical state space formalati for the speech enhancement problem. Based on this formalati, Section 3 presents a H1 ltering algorithm. In section 4, we investigate the performance of the H1 lter for the speech enhancement, and compare with that of the Kalman lter. Conclusions and discussions are followed.

2. PROBLEM FORMULATION
Short segments of speech can be represented by the response of an all pole lter which models the vocal tract [1]. The lter is excited by a pulse train separated by the pitch period for voice sounds, or pseudorandom noise for unvoiced sounds. Thus the speech xk (clean speech) is given by

x =
k

X
n j

ax
j

+w

(1)

=1

where n is the number of modeled poles, a0 j s are the tap-gain parameters characterizing the lter and wk is an excitation. If the speech signal is corrupted with a noise signal vk , the observed noisy speech signal sk is described as follows

s =x +v
k k

(2)

The speech generating mechanism (1) and observation process (2) are illustrated in Figure 1. Equations (1)-(2) can be represented by the following state-space

vk wk
speech xk generator

is minimum. The estimation error ek is dened by the equation

+
measurement system

sk

H oo filter

^ xk

e =x
k

x ^

(6)

Figure 1: The noisy speech generating and ltering mechanism. model [2]-[3]

For the state-space model (3)-(4), the Kalman ltering algorithm is given by ^ k = AX ^ k 1 + Kk [sk CAX ^ k 1] X (7) ^ with the initial condition X0 = [0]n1 . The lter gain and error variance equations are

X = AX 1 + Bw s = CX + v with X = [x +1 x +2 ::: x ] 0 0 1 0 ::: B 0 0 1 ::: B : : : ::: B A = B : : : ::: B @ 0 0 0 ::: a a 1 a 2 ::: B = C = [0 0 : : : 0 1]

k k k k k k k n k n k n n n T

Pj
(3) (4) where
k k k T

K P

k k

1
k

0 0 0

: :

0 0 1

: :

a2 a1

1 C C C C C A

K is a Kalman gain vector, P j 1 = E [(X ^ j 1 ) (X X ^ j 1 )] is a priori error covariance matrix, P = X ^ ^ ) is a posteriori error covariance matrix. E [(X X ) (X X The initial condition P0 = [0]  . I is an nn identity matrix. ^ can be obtained by The estimated speech sample x ^ x ^ = CX (11) If the additive noise fv g is a colored Gaussian process, the Kalman
k k k k k k k k k k T k n n k k k k

= P j 1 C [R + CP j = AP 1A + BQB = [I K C ]P j 1
k k k T k k k

k k T

C ]
T

(8) (9) (10)

lter algorithm for such speech estimation is given in [3].

The objective of speech enhancement is to estimate xk given si (0 i k). Previous studies have assumed that both wk and vk are white, uncorrelated Gaussian processes in order to developing enhancement algorithms, i.e. to estimate xk [1]-[4]. However neither the speech nor the noise may be Gaussian. This is because wk could be a pulse train for voiced speech, random noise for unvoiced speech or the modeling error, vk could be any kind of noise. The Gaussian assumptions may provide an estimate which is highly vulnerable to statistical outliers, i.e., a small number of large measurement errors would have a large inuence on the resulting estimate, so that the viability of the algorithms was checked by experiment [3]. In the following, we present an new approach based on the H1 ltering algorithm for speech enhancement, where both wk and vk may not be white or colored Gaussian processes. For comparison, the Kalman ltering algorithm is briey reviewed.

3.2.

f g

 

H1 Filtering Algorithm

Consider the state space model (3)-(4), we make no assumptions on the nature of unknown quantities wk and vk , and are interested not necessarily in the estimation of Xk but xk using the observations si (0 i k). Let zk = CXk (12) Different from that of the Kalman lter which minimizes the variance of the estimation error, the design criterion for the H1 lter ^k , for any is to provide an uniformly small estimation error, ek =zk -z n wk , vk l2 and X0 . Note that zk is equal to xk . The measure of performance is then given by

f g  

J=

jX0

P jz z ^ j2 =0 P ^ 0 j2 + =0 fjw j2 X
N k k p

2R

k Q

3. 3.1.

KALMAN AND 1 FILTERING ALGORITHMS

^ 0 is an a priori estimate of X0 , ^ 0 ); w ; v ) 6= 0, X where ((X0 X Q  0, p0 1 > 0, W > 0 and V > 0 are the weighting matrices,
k k

N k

+ jv j2 1 g
k V

(13)

Kalman Filtering Algorithm

In the Kalman ltering, the clean speech signal xk is considered to be a random process. Assuming that both exciting term wk and observation noise vk (additive noise) are white Gaussian processes with zero mean and uncorrelated variances Q and R

f g

and are left to the choice of the designer and depend on performance requirements. The notation zk 2 Q is dened as the square of T the weighted (by Q) L2 norm of zk , i.e., zk 2 Q = zk Qzk . The H1 lter will search z ^k such that the optimal estimate zk among all possible z ^k (i.e. the worse-case performance measure) should satisfy

j j

j j

E means expectation. The design objective of Kalman lter is to ^ based on the fs g (0  i  k) determine the optimal estimate x
k i

E fw w g = Q E fv v g = R E fw v g = 0:
k T k T k T j k k

where sup stands for supremum and > 0 is a prescribed level of noise attenuation. The H1 ltering can be interpreted as a minimax problem where the estimator strategy z ^k play against the exogenous inputs wk , vk and the initial state X0 . The performance criterion becomes
2 ^ 2 min max J = 1 ^k  k k 0  2 jX0 X0 j 0 1 X + 1 ^ j2 2 (jw j2 1 + jv j2 1 )] 2 =0 [jz z
z v ;w ;X p N k k Q k W k V k

sup J  2

(14)

such that

P = E fe e g
k k T k

(5)

(15)

where min stands for minimization and max maximization. Note that unlike the traditional minimum variance ltering approach(Wiener and/or Kalman ltering), the H1 ltering deals with deterministic disturbances and no a priori knowledge of the noise statistics is required. Since the the observation sk is given, vk can be uniquely determined by (2) once the optimal values of wk ^ k , we can rewrite the and X0 are found, and zk = CXk , z ^k = C X performance criterion (15) as
2 ^ 2 min max J = 1 ^k  k k 0 2 jX0 X0 j 0 1
X s ;w ;X p N k

4. EXPERIMENTAL RESULTS
Both Kalman and H1 ltering algorithms described in Section 3 are applied to speech enhancement as follows. Noisy speech is divided into equal-length segments. Within each segment, we rst estimate the tap-gain parameters aj ; j = 1; :::;n, then lter the noisy speech with the parameters aj . The Kalman and H1 ltering algorithms are initialized only for the rst seg^ 0 = 0, and ment. In our experiment, we choose the state vector X ^ 0 and p0 0. In the subsequent segments, X weight matrix p0 are initialized using the corresponding last values from the previous segment. There exists a trade off in the choice of the length of the segments. Large segments improve the accuracy of the prediction parameters for stationary sounds (e.g. vowels) and short segments improve the accuracy for nonstationary sounds. In our experiment, the segment length used for calculating the parameters aj ; i = 1; 2; : : : ; n is 128 samples, which corresponds to 16 ms (with a sampling frequence of 8 KHz). The order of the all pole lter is 10, which is a commonly used value in linear predictive analysis of speech signal and the input SNR is 5 dB. Two types of noise are used: white noise (stationary) and helicopter noise (nonstationary). The performance of both ltering algorithms is measured in terms of SNR, time domain speech representation, and listening evaluation. In the testing, the order of state space model is set to be equal to the order of the all pole lter. Experiment results obtained for the sentence Woe betide the interviewee if he answered vaguely are shown in the following tables. Table 1 shows the output SNR results for the testing where the weightings are W = 1; V = 3, the attenuation parameter = 1:02, and p0 = 1000. The SNR values throughout this paper are the global signal to noise ratios calculated by

f g

(16)

 = C T QC . In [9]-[10], it has been proved that the followwhere Q ing theorem presents a complete solution to the H1 ltering problem for the state-space model (3)-(4) with the performance criterion (16).
Theorem : Let > 0 be a prescribed level of noise attenuation. Then, there exists an H1 lter for zk if and only if there exists a stabilizing symmetric solution Pk > 0 to the following discretetime Riccati equation

X + 1 2 =0 [jX
k

^ j2 2 (jw j2 X
k Q

k W

+ js

CX j2 1 )]
k V

P +1 = AP A AP C (V + CP C ) 1 CP A +BWB 2 P +1 L (Q 1 + 2 LP +1 L ) 1 LP P0 = (p0 1 + 2 LQL ) 1 : If this is the case, then an H1 lter can be given by ^ ; k = 1; 2; : : : ; N z ^ = CX
k k T k T k T k T T k T k T T k k

(17)
k

+1

(18)

where

is the gain of the H1 lter and given by

k k T k

^ = AX ^ X
k

+ H (s
k

^ CAX

^0 = 0 1 ); X
T

SNR = 10log10 P
(19)

N k

Solving Riccati equation (17) for the solution Pk is not trivial due to its nonlinearity. Applying the following matrix inversion lemma(MIL)

H = AP C (V + CP C )

(20)

A AB (C + B AB ) 1 B A = (A 1 + BC 1 B ) 1 ;
T T T

(21)

(17) can be written as

1 +1

= [A(P 1 + C V 1 C ) 1 A + BWB ]
T T T k

2 C QC
T

so that we can obtain the solution of Pk from (22) recursively.

(22)

Comparing the Kalman ltering algorithm (7)-(10) and the H1 ltering algorithm (17)-(20), we can observe 1) The Kalman ltering algorithm gives the minimum mean-squareerror estimate of Xk based on the si (0 i k);

where N is the total number of samples of each sentence, xk is ^k is the enhanced speech. Tathe clean (noise-free) sequence and z ble 1 gives the performance comparison between the Kalman and H1 ltering algorithms for the situations that the speech signal is contaminated by white Gaussian noise and helicopter noise respectively. In both cases, the H1 ltering algorithm has over 1.0 dB advantage in the output SNR values over the Kalman ltering algorithm [3]. Figure 2 shows the clean speech signal. Figures 3 and 4 show the original noisy speech signals (contaminated by white Gaussian noise and by helicopter noise respectively) as well as the corresponding noise suppressed signals in the time domain by using the H1 ltering algorithm. Informal subjective tests conrm the improvement of SNR values, intelligibility and voice quality. The gures clearly show the effectiveness of the H1 ltering for speech enhancement.

x2 z ^ ]2 =1 [x
N k

=1
k

(23)

f g  

2) The H1 ltering algorithm gives the optimal estimate of xk based on the si (0 i k) such that the effect of the worst disturbance(noises) on the estimation error is minimized.

f g

 

Table 1: Performance Comparison with Input SNR=5 dB Filtering Output SNR (dB) Algorithm White Noise Helicopter Noise Kalman 8.7276 8.9119 H1 9.8781 10.0693

4000

3000

2000

4000 noisy signal 2000

Amplitude
x 10
4

1000

Amplitude

1000

2000

0 2000 4000 0

3000

4000 0

0.5

1 Sample Number

1.5

Figure 2: The clean speech signal.

0.5

5. CONCLUSIONS
A new speech enhancement method based on the ltering has been developed. This method exploits a speech production model without requiring the knowledge of noise statistics. The effectiveness of the method has been demonstrated based on measurement data and computer simulations. Since the design criterion is based ltering approach is less senon the worst case disturbance, the sitive to uncertainty in the exogenous signal statistics and system model dynamics.

1 Sample Number

1.5

2 x 10
4

H1

4000 filtered signal 2000

Amplitude

H1

0 2000 4000 0

6.

REFERENCES

0.5

1 Sample Number

1.5

2 x 10
4

1. J.S. Lim and A.V. Oppenheim, All-Pole Modeling of Degraded Speech, IEEE Trans. Acoust., Speech, Signal Processing, Vol. ASSP-26, 197-210, 1978. 2. K.K. Paliwal and A. Basu, A Speech Enhancement Method Based on Kalman Filtering, Proc. IEEE ICASSP, 177-180, 1987. 3. J.D. Gibson, B. Koo and S.D. Gray, Filtering of Colored Noise for Speech Enhancement and Coding, IEEE Trans. Signal Processing, Vol. SP-39, 1732-1742, 1991. 4. Y. Ephraim, A Minimum Mean Square Error Approach for Speech Enhancement, Proc. IEEE ICASSP, 829-832, 1990. 5. R.N. Banavar and J.L. Speyer, A Linear Quadratic Game Theory Approach to Estimation and Smoothing, Proc. IEEE ACC, 2818-2822, 1991. Optimal 6. M.J. Grimble and A. Elsayed, Solution of the Linear Filtering Problem for Discrete-Time Systems, IEEE Trans. Acoust., Speech, Signal Processing, Vol. ASSP-38, 1092-1104, 1990.

Figure 3: The white Gaussian noise contaminated speech signal ltering. and the estimated speech signal using the

H1

4000 noisy signal 2000

Amplitude

0 2000 4000 0

H1

0.5

1 Sample Number

1.5

2 x 10
4

4000 filtered signal 2000

Amplitude

-Optimal Estimation: A Tuto7. U. Shaked and Y. Theodor, rial, Proc. 31st IEEE CDC, 2278-2286, 1992. 8. K.M. Nagpal and P.P. Khargonekar, Filtering and Smoothing Setting, IEEE Trans. Auto. Control, Vol. AC-36, in an 152-166, 1991.

H1

H1

0 2000 4000 0

9. X. Shen and L. Deng, Discrete Filter Design with Application to Speech Enhancement, Proc. IEEE ICASSP95, Detroit, 1504-1507, 1995. 10. B. Hassibi and T. Kailath, Adaptive Filtering, Proc. IEEE ICASSP95, Detroit, 949-952, 1995.

H1

0.5

1 Sample Number

1.5

2 x 10
4

H1

Figure 4: The helicopter noise contaminated speech signal and the ltering. estimated speech signal using the

H1