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Available online at www.sciencedirect.com Speech Communication 78 (2016) 73–83 www.elsevier.com/locate/specom Single-channel noise reduction via semi-orthogonal transformations and reduced-rank filtering Wei Zhang a,, Jacob Benesty b , Jingdong Chen a a Center of Intelligent Acoustics and Immersive Communications and School of Marine Science and Technology, Northwestern Polytechnical University, 127 Youyi West Road, Xi’an, Shaanxi 710072, China b INRS-EMT, University of Quebec, 800 de la Gauchetiere Ouest, Suite 6900, Montreal, QC H5A 1K6, Canada Received 10 January 2015; received in revised form 22 November 2015; accepted 2 December 2015 Available online 9 February 2016 Abstract This paper investigates the problem of single-channel noise reduction in the time domain. The objective is to find a lower dimensional filter that can yield a noise reduction performance as close as possible to or even better than that obtained by the full-rank solution. This is achieved in three steps. First, we transform the observation signal vector sequence, through a semi-orthogonal matrix, into a sequence of transformed signal vectors with a reduced dimension. Second, a reduced-rank filter is applied to get an estimate of the clean speech in the transformed domain. Third, the estimate of the clean speech in the time domain is obtained by an inverse semi-orthogonal transformation. The focus of this paper is on the derivation of semi-orthogonal transformations under certain estimation criteria in the first step and the design of the reduced-rank optimal filters that can be used in the second step. We show how noise reduction using the principle of rank reduction can be cast as an optimal filtering problem, and how different semi-orthogonal transformations affect the noise reduction performance. Simulations are performed under various conditions to validate the deduced filters for noise reduction. © 2016 Elsevier B.V. All rights reserved. Keywords: Noise reduction; Speech enhancement; Semi-orthogonal transformation; Wiener filter; MVDR filter; Reduced-rank filter. 1. Introduction The problem of single-channel noise reduction is to recover a clean speech signal of interest from its microphone observa- tions (Benesty and Chen, 2011; Benesty et al., 2009; Loizou, 2007). Due to the importance and broad range of applications, a great deal of efforts have been devoted to this problem over the last decades and many algorithms have been developed e.g., Wiener (1949), Boll (1979), Berouti et al. (1979), Lim and Oppenheim (1979), Ephraim and Malah (1984), Trees and Harry (2001). However, these algorithms achieve noise reduction generally by paying a price of adding speech dis- tortion. One exceptional case is the reduced-rank or subspace This work was supported in part by the NSFC under Grant No. 61425005. Corresponding author. Tel.: +86 02988491079. E-mail addresses: [email protected] (W. Zhang), [email protected] (J. Benesty), [email protected] (J. Chen). method, which has the potential to introduce less distortion if the desired signal correlation matrix is rank deficient and this rank is correctly estimated. This paper is, therefore, devoted to the reduced-rank filtering methods. The idea of “reduced rank” was first developed in the field of signal estimation (Huffel, 1993; Moor, 1993; Scharf, 1991; Scharf and Tufts, 1987; Tufts and Kumaresan, 1982a, 1982b). It was then applied to the noise reduction problem in the so- called subspace approach (Dendrinos et al., 1991), where the singular value decomposition (SVD) of the noisy data ma- trix was used to estimate and remove the noise subspace and the estimate of the clean signal was then obtained from the remaining subspace. This approach gained more popularity when Ephraim and Van Trees proposed to decompose the covariance matrix of the noisy observation vector (Ephraim and Trees, 1995). The subspace method was found better than the widely used spectral subtraction (Boll, 1979) for noise reduction in the sense that it has less speech distortion with little music residual noise. Today, the principle has been http://dx.doi.org/10.1016/j.specom.2015.12.007 0167-6393/© 2016 Elsevier B.V. All rights reserved.

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Available online at www.sciencedirect.com

Speech Communication 78 (2016) 73–83 www.elsevier.com/locate/specom

Single-channel noise reduction via semi-orthogonal transformations and

reduced-rank filtering

Wei Zhang

a , ∗, Jacob Benesty

b , Jingdong Chen

a

a Center of Intelligent Acoustics and Immersive Communications and School of Marine Science and Technology, Northwestern Polytechnical University, 127 Youyi West Road, Xi’an, Shaanxi 710072, China

b INRS-EMT, University of Quebec, 800 de la Gauchetiere Ouest, Suite 6900, Montreal, QC H5A 1K6, Canada

Received 10 January 2015; received in revised form 22 November 2015; accepted 2 December 2015 Available online 9 February 2016

Abstract

This paper investigates the problem of single-channel noise reduction in the time domain. The objective is to find a lower dimensional filter that can yield a noise reduction performance as close as possible to or even better than that obtained by the full-rank solution. This is achieved in three steps. First, we transform the observation signal vector sequence, through a semi-orthogonal matrix, into a sequence of transformed signal vectors with a reduced dimension. Second, a reduced-rank filter is applied to get an estimate of the clean speech in the transformed domain. Third, the estimate of the clean speech in the time domain is obtained by an inverse semi-orthogonal transformation. The focus of this paper is on the derivation of semi-orthogonal transformations under certain estimation criteria in the first step and the design of the reduced-rank optimal filters that can be used in the second step. We show how noise reduction using the principle of rank reduction can be cast as an optimal filtering problem, and how different semi-orthogonal transformations affect the noise reduction performance. Simulations are performed under various conditions to validate the deduced filters for noise reduction. © 2016 Elsevier B.V. All rights reserved.

Keywords: Noise reduction; Speech enhancement; Semi-orthogonal transformation; Wiener filter; MVDR filter; Reduced-rank filter.

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. Introduction

The problem of single-channel noise reduction is to recover clean speech signal of interest from its microphone observa-ions ( Benesty and Chen, 2011; Benesty et al., 2009; Loizou,007 ). Due to the importance and broad range of applications, great deal of efforts have been devoted to this problem overhe last decades and many algorithms have been developed.g., Wiener (1949) , Boll (1979) , Berouti et al. (1979) , Limnd Oppenheim (1979) , Ephraim and Malah (1984) , Treesnd Harry (2001) . However, these algorithms achieve noiseeduction generally by paying a price of adding speech dis-ortion. One exceptional case is the reduced-rank or subspace

� This work was supported in part by the NSFC under Grant No. 1425005 . ∗ Corresponding author. Tel.: +86 02988491079.

E-mail addresses: [email protected] (W. Zhang), [email protected] (J. Benesty), [email protected] (J. Chen).

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ttp://dx.doi.org/10.1016/j.specom.2015.12.007 167-6393/© 2016 Elsevier B.V. All rights reserved.

ethod, which has the potential to introduce less distortion ifhe desired signal correlation matrix is rank deficient and thisank is correctly estimated. This paper is, therefore, devotedo the reduced-rank filtering methods.

The idea of “reduced rank” was first developed in the fieldf signal estimation ( Huffel, 1993; Moor, 1993; Scharf, 1991;charf and Tufts, 1987; Tufts and Kumaresan, 1982a, 1982b ).t was then applied to the noise reduction problem in the so-alled subspace approach ( Dendrinos et al., 1991 ), where theingular value decomposition (SVD) of the noisy data ma-rix was used to estimate and remove the noise subspace andhe estimate of the clean signal was then obtained from theemaining subspace. This approach gained more popularityhen Ephraim and Van Trees proposed to decompose the

ovariance matrix of the noisy observation vector ( Ephraimnd Trees, 1995 ). The subspace method was found betterhan the widely used spectral subtraction ( Boll, 1979 ) foroise reduction in the sense that it has less speech distortionith little music residual noise. Today, the principle has been

74 W. Zhang et al. / Speech Communication 78 (2016) 73–83

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studied to deal with not only white ( Ephraim and Trees, 1995 )but also colored noise ( Hu and Loizou, 2003; Huang andZhao, 1998; 2000; Mittal and Phamdo, 2000; Rezayee andGazor, 2001 ). Besides the SVD ( Moor, 1993; Scharf, 1991;Scharf and Tufts, 1987; Tufts and Kumaresan, 1982a, 1982b )and the eigenvalue decomposition (EVD) ( Ephraim and Trees,1995; Hu and Loizou, 2003 ), truncated (Q)SVD ( Hansen andJensen, 1998; Jensen et al., 1995 ) and triangular decompo-sitions ( Hansen and Jensen, 2007 ) were also investigated inthe subspace approach. More recent works on reduced-rankfiltering can be found in Hansen and Jensen (2013) , Nørholmet al. (2014) , Zhang et al. (2014) .

This paper is also concerned with the application ofreduced-rank principle to noise reduction. But unlike most ex-isting work (e.g., Dendrinos et al., 1991; Ephraim and Trees,1995; Goldstein et al., 1999; 1998; Scharf, 1991; Scharf andTufts, 1987 ), which exploits the structure of either the sig-nal data or covariance matrix to find the signal and noisesubspaces, this paper develops a more flexible framework.We choose a semi-orthogonal matrix to do data transforma-tion instead of directly decomposing the subspaces. The semi-orthogonal matrix is not unique, and it can be derived underdifferent criteria. The resulting semi-orthogonal matrices rep-resent the characteristic of both the signal and noise, and thusmight be used in various conditions. Another contribution ofthe paper is the derivation of the optimal filters under thereduced-rank framework.

In this framework, noise reduction is achieved in threesteps. We first prefilter the full-length observed vector bya semi-orthogonal matrix, resulting in a reduced-dimensionvector. In other words, we apply a linear transformation thattransforms the observed data vector to a new coordinate sys-tem where the basis are defined by the columns of the semi-orthogonal matrix. This is workable because the dimensionof the signal subspace is smaller than that of the observednoisy signal space. The second step is to design an optimalreduced-rank filter and apply this filter to get an estimate ofthe clean speech in the transformed domain. Note that theoptimal filter is matrix-valued and the noisy signal is pro-cessed by a vector-by-vector basis. The estimate of the cleanspeech in the time domain is finally obtained by an inversesemi-orthogonal transformation. We will discuss how to de-rive different semi-orthogonal transformations under certainestimation criteria and how to design different reduced-rankoptimal filters. We will also illustrate the flexibility of thisnew framework in controlling the compromise between noisereduction and speech distortion.

The rest of the paper is organized as follows. In Section 2 ,the signal model and problem formulation are presented. Sec-tion 3 gives the definition of the semi-orthogonal transforma-tion. Then in Section 4 , the principle of linear filtering witha rectangular matrix is discussed. Section 5 presents someperformance measures for evaluation and analysis of noisereduction. In Section 6 , different optimal filters are derivedunder a given semi-orthogonal transformation. Different semi-orthogonal transformations are discussed in Section 7 . Some

imulations are presented in Section 8 . Finally, conclusionsre drawn in Section 9 .

. Signal model and problem formulation

The noise reduction problem considered in this paper isne of recovering the desired speech signal x ( k ), k being theiscrete-time index, of zero mean from the noisy observa-ion (sensor signal) ( Benesty and Chen, 2011; Benesty et al.,009 ):

(k) = x(k) + v(k) , (1)

here v ( k ), assumed to be a zero-mean random process, is thenwanted additive noise that can be either white or coloredut is uncorrelated with x ( k ). All signals are considered to beeal and broadband.

The signal model given in (1) can be put into a vectororm by considering L most recent successive time samples,.e.,

(k) = x(k) + v(k) , (2)

here

(k) � =

[y(k) y(k − 1) · · · y(k − L + 1)

]T (3)

s a vector of length L , superscript T denotes transpose of aector or a matrix, and x ( k ) and v ( k ) are defined in a similaray to y ( k ). Since x ( k ) and v ( k ) are uncorrelated by assump-

ion, the correlation matrix (of size L × L ) of the noisy signalan be written as

y � = E

[y(k) y

T (k) ] = R x + R v , (4)

here E [ ·] denotes mathematical expectation, and R x � =

[x (k) x

T (k) ]

and R v � = E

[v (k) v

T (k) ]

are the correlationatrices of x ( k ) and v ( k ), respectively. The noise correla-

ion matrix, R v , is assumed to be full rank, i.e., equal to . Then, the objective of noise reduction in this paper is tond a “good” estimate of the vector x ( k ) from the observa-

ion signal vector y ( k ) in the sense that the additive noises significantly reduced while the desired signal is not muchistorted.

. Semi-orthogonal transformation

We recall that x ( k ) is the desired signal vector that weant to estimate from the observation signal vector, y ( k ). Let

=

[t 0 t 1 · · · t P−1

](5)

e a semi-orthogonal matrix of size L × P , i.e., T

T T = I P ,here I P is the P × P identity matrix and P ≤ L . We define

he transformed desired signal vector of length P as

′ (k) � = T

T x(k) (6)

=

[x ′ 0 (k) x ′ 1 (k) · · · x ′ P−1 (k)

]T ,

W. Zhang et al. / Speech Communication 78 (2016) 73–83 75

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here x ′ p (k) = t T p x(k) . Now, instead of estimating the vector ( k ) of length L , we will estimate the shorter vector x

′ ( k ) ofength P .

In (6) , x

′ ( k ) is expressed as a function of x ( k ). It is also ofnterest to write x ( k ) as a function of x

′ ( k ). For that, we canecompose x ( k ) into two uncorrelated components ( Benestynd Chen, 2011 ):

(k) = R x T

(T

T R x T

)−1 x

′ (k) + x u (k)

= Px(k) + ( I L − P ) x(k)

= x c (k) + x u (k) , (7)

here I L is the L × L identity matrix, x c ( k ) and x u ( k ) areorrelated and uncorrelated with x

′ ( k ), respectively,

� = R x T

(T

T R x T

)−1 T

T (8)

s a projection matrix of rank P , and

[x c (k) x

T u (k)

] = 0 L×L . (9)

deally, we would like to have x u (k) = 0 L×1 in order to betterecover the L desired signal samples from the estimation ofhe P transformed samples only, but this is not possible ineneral. The best we can do is to find the semi-orthogonalransformation, T , in such a way that the energy of x u ( k ) isinimized. The correlation matrix of x u ( k ) is

x u = E

[x u (k) x

T u (k)

] = ( I L − P ) R x , (10)

o the energy of x u ( k ) is

σ 2 x u = tr ( R x ) − tr ( PR x ) , (11)

here tr ( ·) denotes the trace of a square matrix. Therefore, theinimization of (11) is equivalent to the maximization of the

econd term on the right-hand side of (11) . The term tr( PR x )s maximized when the columns of T , i.e., t 0 , t 1 , . . . , t P−1 , arehe eigenvectors corresponding to the P largest eigenvalues,0 , λ1 , . . . , λP−1 , of R x . As a result, (11) simplifies to

σ 2 x u =

L−1 ∑

i= P

λi , (12)

here λP , λP+1 , . . . , λL−1 are the L − P smallest eigenvaluesf R x . In practice, though, some other semi-orthogonal ma-rices can be used depending on what we want to achieve.

. Linear filtering with a rectangular matrix

In the general linear filtering approach, we estimate theransformed desired signal vector, x

′ ( k ), by applying a linearransformation to y ( k ), i.e.,

′ (k) = H

′ y(k)

= H

′ [ x(k) + v(k) ]

= x

′ fd (k) + v

′ rn (k) , (13)

here z ′ ( k ) is the estimate of x

′ ( k ), H

′ is a rectangular filteringatrix of size P × L ,

′ fd (k)

� = H

′ x(k) (14)

s the filtered desired signal, and

′ rn (k)

� = H

′ v(k) (15)

s the residual noise. Therefore, the estimate of x ( k ) is

(k) = Tz ′ (k)

= TH

′ y(k)

= Hy(k) , (16)

here H = TH

′ is a filtering matrix of size L × L . We find that the correlation matrix of z ′ ( k ) is

z ′ = H

′ R y H

′ T

= H

′ R x H

′ T + H

′ R v H

′ T . (17)

e deduce that tr ( R z ′ ) = tr ( R z ) , where R z = HR y H

T is theorrelation matrix of z ( k ).

. Performance measures

In this section, we define two categories of performanceeasures. The first category evaluates the noise reduction per-

ormance while the second one evaluates the signal distor-ion. We also discuss the very convenient mean-squared errorMSE) criterion and show how it is related to the performanceeasures.

.1. Noise reduction

The most important measure in noise reduction is theignal-to-noise ratio (SNR). The input SNR is defined as

SNR

� =

tr ( R x )

tr ( R v ) =

σ 2 x

σ 2 v

, (18)

here σ 2 x

� = E

[x 2 (k)

]and σ 2

v � = E

[v 2 (k)

]are the variances

f x ( k ) and v ( k ), respectively. The output SNR, obtained from17) , helps quantify the SNR after filtering. It is given by

SNR

(H

′ ) � =

tr (H

′ R x H

′ T )

tr (H

′ R v H

′ T ) = oSNR ( H ) . (19)

he objective of noise reduction is to find an appropriate

′ that will make the output SNR greater than the inputNR. Consequently, the quality of the noisy signal may benhanced.

The noise reduction factor quantifies the amount of noiseeing rejected by H

′ . This quantity is defined as the ratio ofhe power of the noise at the sensor over the power of theoise remaining after filtering, i.e.,

nr (H

′ ) � =

tr ( R v )

tr (H

′ R v H

′ T ) = ξnr ( H ) . (20)

ny good choice of H

′ should lead to ξ nr ( H

′ ) ≥ 1.

76 W. Zhang et al. / Speech Communication 78 (2016) 73–83

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5.2. Speech distortion

The transformed desired speech signal can be distorted bythe rectangular filtering matrix. Therefore, the speech reduc-tion factor is defined as

ξsr (H

′ ) � =

tr ( R x )

tr (H

′ R x H

′ T ) = ξsr ( H ) . (21)

We should have ξ sr ( H

′ ) ≥ 1. A rectangular filtering matrix that does not affect the trans-

formed desired signal vector, x

′ ( k ), requires the constraint:

H

′ R x T

(T

T R x T

)−1 = H

′ PT = I P . (22)

By making the appropriate substitutions, one can derivethe relationship among the measures defined so far, i.e.,

oSNR

(H

′ )iSNR

=

ξnr (H

′ )ξsr ( H

′ ) . (23)

Another way to measure the distortion of the transformeddesired signal due to the filtering operation is via the speechdistortion index defined as

υsd (H

′ ) =

E

{ [x

′ fd (k) − x

′ (k) ]T [

x

′ fd (k) − x

′ (k) ]}

tr ( R x ) (24)

= υsd ( H ) .

The speech distortion index is always greater than or equal to0; the higher is the value of υsd

(H

′ ), the more the transformeddesired signal is distorted.

Besides the above performance measures, the perceptualevaluation of speech quality (PESQ) ( ITU ) is also used inour simulations and experiments.

5.3. MSE criterion

The transformed desired signal is a vector of length P ,and so is the error signal. We define the error signal vectorbetween the estimated and desired signals as

e ′ (k) � = z ′ (k) − x

′ (k) (25)

= H

′ y(k) − x

′ (k) ,

which can also be expressed as the sum of two orthogonalerror signal vectors:

e ′ (k) = e ′ ds (k) + e ′ rs (k) , (26)

where

e ′ ds (k) � = x

′ fd (k) − x

′ (k)

=

(H

′ − T

T )x(k) (27)

is the signal distortion due to the rectangular filtering matrixand

e ′ rs (k) � = v

′ rn (k) = H

′ v(k) (28)

represents the residual noise. Therefore, the MSE criterionis

(H

′ ) � = tr {E

[e ′ (k) e ′ T (k)

]}(29)

= tr (T

T R x T

) + tr (H

′ R y H

′ T ) − 2 tr (H

′ R x T

).

sing the fact that E

[e ′ ds (k) e ′ T rs (k)

] = 0 P×P , J ( H

′ ) can be ex-ressed as the sum of two other MSEs, i.e.,

(H

′ ) = tr {E

[e ′ ds (k) e ′ T ds (k)

]} + tr {E

[e ′ rs (k) e ′ T rs (k)

]}

= J ds (H

′ ) + J rs (H

′ ), (30)

here

ds (H

′ ) � = tr [ (

H

′ − T

T )R x

(H

′ − T

T )T

]

= tr ( R x ) υsd (H

′ ) (31)

nd

rs (H

′ ) � = tr (H

′ R v H

′ T )

=

tr ( R v )

ξnr ( H

′ ) . (32)

e deduce that

J ds (H

′ )J rs ( H

′ ) = iSNR × ξnr

(H

′ ) × υsd (H

′ )

= oSNR

(H

′ ) × ξsr (H

′ ) × υsd (H

′ ). (33)

e observe how the MSEs are related to the performanceeasures.

. Optimal rectangular filtering matrices

In this section, we briefly discuss the most important opti-al rectangular filtering matrices for noise reduction, which

xplicitly depend on the semi-orthogonal matrix T .

.1. Maximum SNR

It can be shown that the maximum SNR filtering matrix isiven by Benesty and Chen (2011)

′ max =

⎢ ⎢ ⎢ ⎣

ς 0 b

T max

ς 1 b

T max . . .

ς P−1 b

T max

⎥ ⎥ ⎥ ⎦

, (34)

here ς p , p = 0, 1 , . . . , P − 1 are arbitrary real numbersith at least one of them different from 0 and b max is the

igenvector corresponding to the maximum eigenvalue, λmax ,f the matrix R

−1 v R x . Furthermore, we have

SNR

(H

′ max

) = λmax ≥ iSNR (35)

nd

≤ oSNR

(H

′ ) ≤ oSNR

(H

′ max

), ∀ H

′ . (36)

The ς p ’s are found in such a way that distortion of theesired signal is minimized. We obtain

p =

b

T max R x t p λmax

, p = 0, 1 , . . . , P − 1 . (37)

W. Zhang et al. / Speech Communication 78 (2016) 73–83 77

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ubstituting these optimal values into (34) , we obtain the opti-al maximum SNR filtering matrix with minimum distortion

o the transformed desired signal:

′ max = T

T R x b max b

T max

λmax . (38)

e also deduce that the maximum SNR filtering matrix forhe estimation of x ( k ) is

max = TT

T R x b max b

T max

λmax . (39)

.2. Wiener

If we differentiate the MSE criterion, J ( H

′ ), with respecto H

′ and equate the result to zero, we find the Wiener rect-ngular filtering matrix:

′ W

= T

T R x R

−1 y

= T

T (I L − R v R

−1 y

). (40)

e deduce that the Wiener square filtering matrix for thestimation of the vector x ( k ) is

W

= TH

′ W

= TT

T (I L − R v R

−1 y

), (41)

hich does not correspond, in general, to the classical Wienerltering matrix ( Benesty et al., 2009 ). But for P = L, (41)ecomes the well-known Wiener filter:

W

= I L − R v R

−1 y . (42)

Obviously, we have

SNR

(H

′ W

) ≤ oSNR

(H

′ max

)(43)

nd

sd (H

′ W

) ≤ υsd (H

′ max

). (44)

.3. Minimum variance distortionless response

It is possible to derive the minimum variance distortion-ess response (MVDR) filter [distortionless in the sense that

′ ( k ) is left intact in the filtering process] by minimizing theSE of the residual noise, J rs ( H

′ ), with the constraint thathe transformed desired signal, x

′ ( k ), is not distorted, see (22) .athematically, this is equivalent to

in

H

′ tr (H

′ R v H

′ T ) subject to H

′ PT = I P . (45)

he solution to the above optimization problem is

′ MVDR

=

(T

T P

T R

−1 v PT

)−1 T

T P

T R

−1 v . (46)

e deduce that the MVDR for the estimation of x ( k ) is

MVDR

= T

(T

T P

T R

−1 v PT

)−1 T

T P

T R

−1 v . (47)

f course, for P = L, the MVDR filtering matrix degenerateso the identity matrix, i.e., H MVDR

= I L .

We should have

SNR

(H

′ MVDR

) ≤ oSNR

(H

′ W

)(48)

nd

sd (H

′ MVDR

) ≤ υsd (H

′ W

). (49)

Another possible MVDR filtering matrix for the estimationf x ( k ) can be derived through

in

H

′ tr (H

′ R y H

′ T ) subject to H

′ PT = I P , (50)

nd the corresponding solution is

MVDR , 2 = T

(T

T P

T R

−1 y PT

)−1 T

T P

T R

−1 y . (51)

.4. Tradeoff

In the tradeoff approach, we minimize the speech distortionndex with the constraint that the noise reduction factor isqual to a positive value that is greater than 1. Mathematically,his is equivalent to

in

H

′ J ds

(H

′ ) subject to J rs (H

′ ) = βtr ( R v ) , (52)

here 0 < β < 1 to insure that we get some noise reduc-ion. By using a Lagrange multiplier, μ > 0, to adjoin theonstraint to the cost function, we easily deduce the tradeoffltering matrix:

′ T,μ = T

T R x ( R x + μR v ) −1 , (53)

hich can be rewritten, for the estimation of x ( k ), as

T,μ = TT

T R x ( R x + μR v ) −1 . (54)

or P = L, (54) degenerates to the classical tradeoff filter:

T,μ = R x ( R x + μR v ) −1 . (55)

Usually, μ is chosen in a heuristic way, so that for

• μ = 1 , we have H

′ T, 1 = H

′ W

, which is the Wiener filteringmatrix;

• μ > 1, we obtain a filtering matrix with low residual noiseat the expense of high transformed desired signal distortion(as compared to Wiener), and

• μ < 1, we obtain a filtering matrix with high residual noiseand low transformed desired signal distortion (as comparedto Wiener).

. Examples of semi-orthogonal matrices

In Section 3 , we gave an example for the choice of T .n this section, we show some other important possibilitiesepending on what we want to achieve.

78 W. Zhang et al. / Speech Communication 78 (2016) 73–83

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7.1. Minimum MSE

Let us take the Wiener filtering matrix [ Eq. (40) ]. Theminimum MSE (MMSE) is

J (H

′ W

) = tr (T

T R x T

) − tr (T

T R x R

−1 y R x T

). (56)

This MMSE, of course, depends on T . It is easy to check thatthe smallest value for (56) is obtained when the columns of Tare the eigenvectors corresponding to the P largest eigenvaluesof R 1 = R x R

−1 y R x . We denote this semi-orthogonal matrix

by T 1 . In this case, we obtain the well-known reduced-rankWiener filtering matrix ( Goldstein and Reed, 1997; Scharf,1991; Scharf and Tufts, 1987 ):

H W

= T 1 T

T 1 R x R

−1 y . (57)

We also deduce the other reduced-rank optimal filtering ma-trices:

H max = T 1 T

T 1 R x

b max b

T max

λmax , (58)

H MVDR

= T 1 (T

T 1 P

T R

−1 v PT 1

)−1 T

T 1 P

T R

−1 v , (59)

and

H T,μ = T 1 T

T 1 R x ( R x + μR v )

−1 . (60)

7.2. Minimum distortion

The distortion-based MSE with Wiener is

J ds (H

′ W

) = tr (T

T R x T

) − tr (T

T R 2 T

), (61)

where

R 2 = 2R 1 − R x R

−1 y R 1 . (62)

One can verify that the smallest value for (61) is obtainedwhen the columns of T are the eigenvectors correspond-ing to the P largest eigenvalues of R 2 . Let us denote byT 2 this semi-orthogonal matrix. By simply replacing T 2

by T 1 in (57) –(60) , we obtain the new optimal filteringmatrices.

7.3. Minimum residual noise

Let us consider, again, the Wiener filtering ma-trix. The residual noise corresponding to this optimalmatrix is

J rs (H

′ W

) = tr (T

T R 3 T

), (63)

where

R 3 = R x R

−1 y R v R

−1 y R x . (64)

The smallest value for (63) is found when the columns of Tare the eigenvectors corresponding to the P smallest eigenval-ues of R 3 . If we denote by T 3 this semi-orthogonal matrix andreplacing T 3 by T 1 in (57) –(60) , we obtain the new optimalfiltering matrices. This approach may lead to large distortions.

. Simulations and experiments

In this section, we study, using simulations and exper-ments, the impact of some important parameters on theoise reduction performance of the optimal filters derived inection 6 .

.1. Clean speech signal and noise

The clean speech signal used in most of our simulations isecorded from a female speaker in a quiet office room. It isampled at 8 kHz. The overall length of the signal is approx-mately 30-s long. The noisy signal is generated by addingoise to the desired signal, where noise signal is properlycaled to control the input SNR. We study three differentypes of noise, i.e., white Gaussian, car, and babble noise,hich are representative noise samples from white and sta-

ionary to highly nonstationary. The car noise is recorded from sedan car running at 50 miles/hour on a highway with allhe windows closed; this noise is still close to stationary, but its colored. The babble noise is recorded in a New York Stockxchange (NYSE) room; it consists of sounds from variousources such as speakers, telephone rings, electric fans, etc.nd is highly nonstationary.

We also present a set of experiments with real recordedignals where a pre-recorded speech signal is played backhrough a loudspeaker in a noisy but light reverberantoom and we use a microphone to record the noisyignal.

.2. Estimation of the signal statistics

The implementation of all the noise reduction filters de-ived in the previous sections requires the estimation of theorrelation matrices R y , R x , and R v . In most of our simula-ions, we compute the R y and R v matrices using the most re-ent 320 samples (40-ms long) of the noisy and noise signals,espectively, with a short-time average at every time instant . An estimate of the R x matrix is then obtained by subtract-ng R v from R y . To ensure that the R x estimate is positiveemi-definite, we replace all the negative eigenvalues of thisstimated matrix with a very small threshold.

.3. Noise estimation

In some of our simulations, we assume that the noiseignal is accessible and its correlation matrix is com-uted with a short time average. This provides a fairay to compare the performance of different noise re-uction filters. In reality, however, the noise is not ac-essible and has to be estimated from the noisy signal.o, in some simulations and experiments, we also adopt

he improved minima controlled recursive averaging (IM-RA) approach ( Cohen, 2003 ), which has been proved toe a robust noise estimator in a broad range of noisenvironments.

W. Zhang et al. / Speech Communication 78 (2016) 73–83 79

◦: Hmax

: HW

×: HMVDR

−−: iSNR : HT,µ, µ = 0.5

∗: HT,µ, µ = 2

oSN

R(d

B)

24

20

16

12

8

◦: Hmax

: HW

×: HMVDR

: HT,µ, µ = 0.5

∗: HT,µ, µ = 2

υsd

(dB

)

0

-10

-20

-30

10 20 30 40 50

◦: Hmax

: HW

×: HMVDR

−−: iPESQ

: HT,µ, µ = 0.5

∗: HT,µ, µ = 2

L

PE

SQ

a

b

c 3.0

2.5

2.0

1.5

1.0

Fig. 1. Performance of the maximum SNR, Wiener, MVDR, and the tradeoff filters as a function of the filter length, L , in white Gaussian noise: (a) output SNR, (b) speech distortion index, and (c) PESQ score. Simulation conditions: iSNR = 10˜ dB and P = 10.

8

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(

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fi

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t

t

S

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m

i

d

μ

◦: Hmax

: HW

×: HMVDR

−−: iSNR

: HT,µ, µ = 0.5

∗: HT,µ, µ = 2oSN

R(d

B)

24

20

16

12

8

y

y

y

y

◦: Hmax

: HW

×: HMVDR

: HT,µ, µ = 0.5

∗: HT,µ, µ = 2

υsd

(dB

)

0

-10

-20

-30

10 12 14 16 18 20 22 24 26 28 30

◦: Hmax

: HW

×: HMVDR

−−: iPESQ

: HT,µ, µ = 0.5

∗: HT,µ, µ = 2

P

PE

SQ

a

b

c 3.0

2.5

2.0

1.5

1.0

Fig. 2. Performance of the maximum SNR, Wiener, MVDR, and tradeoff filters as a function of the rank parameter, P , in white Gaussian noise: (a) output SNR, (b) speech distortion index, and (c) PESQ score. Simulation conditions: iSNR = 10 dB and L = 30.

t

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.4. Performance of optimal rectangular filtering matrices

In this section, we present the evaluation of the noise re-uction filters with three performance measures: the outputNR defined in (19) , the speech distortion index defined in24) , and the PESQ ( ITU ). The semi-orthogonal matrix Tsed here is obtained by minimizing the energy of the sig-al that is uncorrelated with the transformed signal, and itsolumns consists of the eigenvectors corresponding to the Pargest eigenvalues of R x .

Fig. 1 plots the simulation results of the noise reductionlters as a function of the filter length, L . The backgroundoise is white Gaussian, the input SNR is 10 dB, P is seto 10, and the noise statistics are computed directly fromhe noise signal. It is seen that all filters can improve theNR with a similar degree of speech distortion except theor maximum SNR filter. The output SNR and the speechistortion index both increase with the filter length, L . Theaximum SNR filter expectedly has overwhelming superior-

ty in output SNR, as its name indicates; but it also intro-uces tremendous speech distortion. The tradeoff filter with= 2 achieves higher output SNR and speech distortion than

he Wiener filter, while that with μ = 0. 5 behaves the oppo-ite, which is in good agreement with what was analyzed inection 6 . The SNR improvement of the MVDR filter is the

east among all the studied filters, but the value of its speechistortion index is also the smallest. It is also seen that thenfluence of the filter length on the speech distortion indexs dramatic when it is small, but less remarkable as the filterength is large. Finally, the PESQ results indicate that: (1) theest performance generally appears when the filter length islightly larger than the signal dimension P ; (2) all the noiseeduction filters except for the maximum SNR one improvehe quality. The PESQ score of the maximum SNR filter isven worse than the original noisy speech. These results cor-oborate with what was observed in the literature of noiseeduction ( Benesty et al., 2009 ).

Fig. 2 plots the performance of different noise reductionlters as a function of the rank parameter, P , in white Gaus-ian noise. The input SNR is 10 dB and the filter length iset to 30 based on the previous simulation. Again, the noisetatistics are computed directly from the noise signal. It iseen that the output SNR decreases as P increases in thetudied range of P (note that the smallest value of P is set

80 W. Zhang et al. / Speech Communication 78 (2016) 73–83

◦: Hmax

: HW

×: HMVDR

: HT,µ, µ = 0.5

∗: HT,µ, µ = 2

oSN

R(d

B)

a

b

c

36

27

18

9

0

◦: Hmax

: HW

×: HMVDR

: HT,µ, µ = 0.5

∗: HT,µ, µ = 2

υsd

(dB

)

0

-5

-10

-15

-20

-25

0 5 10 15 20 25

◦: Hmax

: HW

×: HMVDR

: HT,µ, µ = 0.5

∗: HT,µ, µ = 2

−−: iPESQ

iSNR (dB)

PE

SQ

3.5

3.0

2.5

2.0

1.5

1.0

Fig. 3. Performance of the maximum SNR, Wiener, MVDR, and tradeoff filters as a function of input SNR in white Gaussian noise: (a) output SNR, (b) speech distortion index, and (c) PESQ score. Simulation conditions: L =

30 and P = 20.

Table 1 Mean Opinion Score (MOS) Standard.

MOS Quality Impairment

5 Excellent Imperceptible 4 Good Perceptible but not annoying 3 Fair Slightly annoying 2 Poor Annoying 1 Bad Very annoying

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d

to 10 in our simulations since we found that the level ofspeech distortion is acceptable with P ≥ 10), and so is thespeech distortion index. The PESQ scores, however, do notchange much with P for all the filters except for the MVDRone. Note that as the value of P approaches the value ofL , the reduced-rank filters approaches the full-rank solutions.So, the results in this simulation show that the reduced-ranknoise reduction filters can yield a similar PESQ performanceas their full-rank counterparts as long as the rank parame-ter, P , is in a reasonable range; but the reduced-rank solutionprovides an extra degree of freedom to compromise betweennoise reduction and speech distortion. For the MVDR filter,more constraint is applied to ensure no speech distortion asthe value of P increases and, as a result, there is less noisereduction. When P = L, the MVDR filter degenerates to theidentity matrix, so there is neither noise reduction nor speechdistortion. This is validated by the results in Fig. 2 .

Simulations were also conducted to evaluate the perfor-mance of the noise reduction filters in different input SNRconditions. Again, the background noise is white Gaussianand the noise statistics are computed directly from the noisesignal. The filter length L = 30 and the rank parameter P =20. The results are plotted in Fig. 3 . In all studied input SNRconditions, the Wiener, tradeoff, and MVDR filters can im-prove both the SNR and the PESQ score. However, the max-imum SNR filter achieves the highest output SNR and itsspeech distortion index does not change much with the inputSNR. Taking into account the previous simulations, we canconclude that the maximum SNR filter should be regarded asthe filter setting an upper bound on the output SNR ratherthan a practical solution in real applications.

In this simulation, we study the performance of differentfilters with the use of the IMCRA method ( Cohen, 2003 )for noise estimation. The filter length L is set to 30, thenoise is white Gaussian and the input SNR is 10 dB. Thecorresponding performance of different noise reduction filtersas a function of the rank parameter, P , is plotted in Fig. 4 .Comparing Figs. 4 and 2 , one can see that the performancewith noise being estimated and that with a priori assumedknown noise are similar. Fig. 5 plots the noisy speech, en-hanced speech, and their spectrograms of the Wiener filterwith L = 30 and P = 14. It can be clearly seen that thereduced-rank Wiener filter with the IMCRA noise estima-tion method achieves tremendous noise reduction, as seenfrom both the waveforms and spectrograms. To further vali-date the noise reduction performance, subjective evaluation isperformed where 18 listeners are asked to rate speech qual-ity according to the mean opinion score (MOS) standard inTable 1 . The 18 listeners are graduate students and profes-sors at Northwestern Polytechnical University. Nine of thoselisteners are familiar with speech quality and speech process-ing while the rest are not familiar with speech evaluation andthey were just taught how to rate MOS scores. The resultsare as follows. The MOS score of the noisy speech is 2.44.With the Wiener filter ( L = 30 and P = 14), the MOS scoreof the enhanced speech is 3.11. The MOS improvement isapproximately 0.7, which is significant.

In this experiment, we study the noise reduction per-ormance with real recorded speech. Since neither noiseor clean speech is accessible in this situation, we use theMCRA method to estimate the noise statistics and subjectivevaluation to assess the quality. The outputs of the Wienerlter with L = 30 and P = 14 for this experiment are plotted

n Fig. 6 . It can be seen that much noise has been reduced.sing the MOS test with 18 listeners, the MOS of the noisy

peech is 2.38, and that for the enhanced speech is 2.72. Oneay notice that the improvement in MOS for real recorded

peech is less than that for the simulated speech. The under-ying reasons are as follows: (1) the noise statistics are moreifficult to estimate and (2) the playback-recording system

W. Zhang et al. / Speech Communication 78 (2016) 73–83 81

◦: Hmax

: HW

×: HMVDR

−−: iSNR

: HT,µ, µ = 0.5

∗: HT,µ, µ = 2oSN

R(d

B)

a

b

c

24

20

16

12

8

◦: Hmax

: HW

×: HMVDR

: HT,µ, µ = 0.5

∗: HT,µ, µ = 2

υsd

(dB

)

0

-10

-20

-30

10 12 14 16 18 20 22 24 26 28 30

◦: Hmax

: HW

×: HMVDR

−−: iPESQ

: HT,µ, µ = 0.5

∗: HT,µ, µ = 2

P

PE

SQ

3.0

2.5

2.0

1.5

1.0

Fig. 4. Performance of the maximum SNR, Wiener, MVDR, and tradeoff filters as a function of the rank parameter, P , with noise estimation using IMCRA: (a) output SNR, (b) speech distortion index, and (c) PESQ score. Simulation conditions: white noise, iSNR = 10˜ dB , and L = 30.

a

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0.5

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quen

cy(k

Hz)

1

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-1

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plitu

de

0 1 2 3 4 5

4

3

2

1

0

1

0.5

0

Time (second)

Fre

quen

cy(k

Hz)

Fig. 5. Outputs of the Wiener filter in white Gaussian noise with noise es- timation using IMCRA: (a) waveform of the noisy signal,(b) spectrogram

of the noisy signal, (c) waveform of the filtered signal, and (d) spectro- gram of the filtered signal. Simulation conditions: L = 30, P = 14, and iSNR = 10 dB .

1

0

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a

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plit

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3

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0.5

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quen

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Hz)

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-1

Am

plit

ude

0 1 2 3 4 5

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3

2

1

0

1

0.5

0

Time (second)

Fre

quen

cy(k

Hz)

Fig. 6. Outputs of the Wiener filter for real recorded signal with noise esti- mation using IMCRA: (a) waveform of the noisy signal, (b) spectrogram of the noisy signal, (c) waveform of the filtered signal, and (d) spectrogram of the filtered signal. Simulation conditions: L = 30 and P = 14.

lso introduces some distortion, which is not additive andannot be efficiently dealt with noise reduction techniques.

.5. Impact of semi-orthogonal transformations on noise eduction performance

Section 7 presented three important semi-orthogonal trans-ormations. They were derived from three different criteria,.e., minimum correlation, minimum MSE (reduced-rank), andinimum distortion. In this section, we study the performance

f the Wiener filter with different semi-orthogonal transforma-ions and compare it to that of the Wiener filter in Section 6 .hree noise environments (white Gaussian, car, and NYSE)re investigated and the rank parameter, P , is set to 10. TheESQ results as a function of the filter length, L , are plotted

n Fig. 7 . Note that the performance of the transformationerived from the minimum-residual-noise criterion was founduch worse than that of the other two transformations. The

nderlying reason is that this criterion does not apply anyonstraint to speech distortion, so the results of this transfor-ation are not plotted in Fig. 7 .

82 W. Zhang et al. / Speech Communication 78 (2016) 73–83

a

b

c

Fig. 7. PESQ scores of the Wiener filter as a function of the filter length, L , in different noise conditions: (a) white Gaussian noise, (b) car noise, and (c) babble noise. Simulation conditions: P = 10 and iSNR = 10˜ dB . The minimum correlation refers to the method in Section 6 , where the semi- orthogonal transformation is constructed using the eigenvectors corresponding to the P largest eigenvalues of R x .

R

B

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G

G

H

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L

M

M

It is seen from Fig. 7 that the PESQ results of the Wienerfilter with the two transformations derived from the minimumMSE and minimum distortion are similar, which is slightlyworse than that of the transformation used in Section 6 whenL is large.

9. Conclusion

This paper investigated the use of the reduced-rank prin-ciple to the problem of single-channel noise reduction inthe framework of semi-orthogonal transformations. Under thisframework, we derived the maximum SNR, the Wiener, theMVDR, and the tradeoff filters. Simulation results showedthat these reduced-rank optimal filters can yield better outputSNR and similar PESQ performance as compared to theirfull-rank counterparts if the rank parameter is properly cho-sen. Furthermore, these reduced-rank filter are more flexiblein controlling the compromise between the amount of noisereduction and the degree of speech distortion than their full-rank counterparts. We also discussed how to derive differ-ent semi-orthogonal transformations though these transforma-tions may not affect much the noise reduction performance interms of PESQ score if the rank parameter is properly cho-sen, but they affect the amount of noise reduction and speechdistortion.

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