International Journal of Advances in Science and Technology,

Vol. 3, No.4, 2011

Audio Signal Enhancement Using Diagonal
Estimator
K.Sreedhar
Department of Electronics and Communication Engineering, VITS (N9), Karimnagar-505001, Andhrapradesh, INDIA
Email: sreedhar_kallem@yahoo.com
Abstract

Audio signals are often contaminated by background environment noise and buzzing or humming noise
from audio equipments. Audio denoising aims at attenuating the noise while retaining the underlying
signals. Removing noise from audio signals requires a diagonal processing of time-frequency coefficients
to avoid producing musical noise. A Log Spectral Amplitude estimation procedure is introduced, which
adjusts all parameters adaptively to signal property by minimizing a Stein estimation of the risk. Diagonal
time-frequency audio denoising algorithm attenuates the noise by processing each spectrogram
coefficient independently. This Estimator is to minimize the error between clean signal and the enhanced
signal.
Keywords: Audio Enhancement, MMSE LSAE, Audio signal processing, STFT Transform, Spectrogram,
Time-Frequency Audio Denoising, Adaptive Block Thresholding

1. Introduction

Recently proposed an algorithm for enhancing speech degraded by uncorrelated additive noise when the
noisy speech alone is available. This algorithm capitalizes on the major importance of the short-time spectral
amplitude of the speech signal in its perception, and utilizes a minimum mean-square error (MMSE)
estimator for enhancing the noisy speech. While the distortion measure of mean-square error of the spectra
used and leads also to good results, it is not the most subjectively meaningful one. It is well known that a
distortion measure which is based on the mean-square error of the log-spectra is more suitable for speech
processing. Such a distortion measure is therefore extensively used for speech analysis and recognition. For
this reason, it is of great interest to examine the estimator which minimizes the mean-square error of the log-
spectra in enhancing noisy speech [1].
2. Problem Statement

The problem of enhancing speech degraded by uncorrelated additive noise, when the noisy speech alone
is available, has recently received much attention. This is due to the many potential applications a successful
speech enhancement system can have, and because of the available technology which enables the
implementation of such intricate algorithms. A comprehensive review of the various speech enhancement
systems which emerged in recent years, and their classification according to the aspects of speech production
and perception which they capitalize on the class of speech enhancement systems which capitalize on the
major importance of the short-time spectral amplitude (STSA) of the speech signal in its perception. In these
systems the STSA of the speech signal is estimated, and combined with the short-time phase of the degraded
speech, for constructing the enhanced signal [2], [3]. The spectral subtraction algorithm and Wiener
filtering are well known examples. In the spectral subtraction algorithm, the STSA is estimated as the
square root of the maximum likelihood (ML) estimator of each signal spectral component variance. In
systems which are based on Wiener filtering, the STSA estimator is obtained as the modulus of the optimal
minimum mean-square error (MMSE) estimator of each signal spectral component. These two STSA
estimators were derived under Gaussian assumption. Since the spectral subtraction STSA estimator is
derived from an optimal (in the ML sense) variance estimator, and the Wiener STSA estimator is derived
from the optimal MMSE signal spectral estimator; both are not optimal spectral amplitude estimators under
the assumed statistical model and criterion. This observation led us to look for an optimal STSA estimator
which is derived directly from the noisy observations. We concentrate here on the derivation of an MMSE
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STSA estimator and on its application in a speech enhancement system [4]. The STSA estimation problem
formulated here is that of estimating the modulus of each complex Fourier expansion coefficient of the
speech signal in a given analysis frame from the noisy speech in that frame. This formulation is motivated by
the fact that the Fourier expansion coefficients of a given signal segment are samples of its Fourier
transform, and by the close relation between the Fourier series expansion and the discrete Fourier transform.
The latter relation enables an efficient implementation of the resulting algorithm by utilizing the FFT
algorithm [5], [6]. To derive the MMSE STSA estimator, the a priori probability distribution of the speech
and noise Fourier expansion coefficients should be known. Since in practice they are unknown, one can
think of measuring each probability distribution or, alternatively, assume a reasonable statistical model. In
the discussed problem, the speech and possibly also the noise are neither stationary nor ergodic processes.
This fact excludes the convenient possibility of obtaining the above probability distributions by examining
the long time behavior of each process. Hence, the only way which can be used is to examine independent
sample functions belonging to the ensemble of each process, e.g., for the speech process these sample
functions can be obtained from different speakers. However, since the probability distributions we are
dealing with are time-varying (due to the nonstationarity of the processes), their measurement and
characterization by the above way is complicated, and the entire procedure seems to be impracticable. For
the above reasons, a statistical model is used here. This model utilizes asymptotic statistical properties of the
Fourier expansion coefficients [7]. In this correspondence we derive a STSA estimator which minimizes the
mean-square error of the log-spectra (i.e., the original STSA and its estimator) and examine it in enhancing
noisy speech [9]. We found that this estimator is superior to the MMSE STSA estimator. Since it results in a
much lower residual noise level without further affecting the speech itself. In fact, the new estimator results
in a very similar enhanced speech quality as that obtained with the MMSE STSA estimator of which takes
into account the signal presence uncertainty.
3. Proposed System

3.1 Diagonal Estimation

Simple time-frequency denoising algorithms compute each attenuation factor only from the
corresponding noisy coefficient and are thus called diagonal estimators [26]. These algorithms have a limited
performance and produce a musical noise. In Diagonal Estimation the Posterior SNR is considered. Posterior
SNR is the SNR of the Audio Noisy Signal.
Diagonal estimators of the SNR ( , ) l k are computed from the a posteriori SNR defined by
2
2
[ , ]
( , )
[ , ]
Y l k
l k
l k

o
=
(1)
One can verify that

[ , ] [ , ] 1 l k l k = is an unbiased estimator.
The empirical Wiener estimator is defined as

1
[ , ] 1

[ , ] 1
( )
a l k
l k +
=
+
(2)
with the notation
max( , 0)
( )
z
z
+
=

Variants of this empirical Wiener are obtained by minimizing a sum of signal distortion and residual
noise energy. The empirical Wiener attenuation rule [28] is given as

2 1 1
[ , ] 1

[ , ] 1
( ) [ ]
a l k
l k
| |

+
=
+
(3)
Where
1 2
, 0 | | >
and 1 > is an over-subtraction factor to compensate variation of noise amplitude.
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The attenuation factor [ , ] a l k of these diagonal estimators only depends upon [ , ] Y l k with no time-
frequency regularization. The resulting attenuated coefficients [ , ] a l k [ , ] Y l k thus lack of time-frequency
regularity. It produces isolated time-frequency coefficients which restore isolated time-frequency structures
that are perceived as a musical noise. A soft thresholding produces a similar phenomenon because each
coefficient is also thresholded independently from its neighbors. To remove this musical noise, uses a block
thresholding estimator that takes into account the fact that large spectrogram coefficients of most audio
sounds are aggregated together in the time-frequency plane [11].

Figure.1. Block diagram of MMSE-LSAE Estimator
Thresholding estimators decompose noisy signals in a basis or in a frame and set to zero small amplitude
coefficients. A diagonal estimator in this basis modifies the amplitude of each coefficient y
f
[m] with a factor
a[m] and reconstructs

f
1

[ ]y [ ]
m
g
N
m
f a m m
=
=

(4)
The estimator is said to be diagonal if a[m] depends only upon y
f
[m].

Figure.2. Block Diagram of Denoising Musical Audio noise signal
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The time-frequency diagonal estimator can be applied directly with short-time Fourier frames. Some
specifications about choice of parameters are discussed below
3.2 Choice of Block

We group time-frequency contiguous short-time Fourier coefficients in disjoint rectangular blocks. The
block size is B
k
#
= L
k
W
k
, where L
k
and W
k
are respectively the block length in time and the block width in
frequency. For simplicity, lengths L
k
= 8, 4, 2 and widths W
k
= 16, 8, 4, 2, 1 will be used (the unit being the
time-frequency index in spectrogram). In this section, fixed block length and width are assigned to all the
blocks, i.e., L
k
= L, W
k
= W and B
#
k
= B
#
= L W. The MMSE amplitude estimator under the assumed
Gaussian statistical model, and uncertainty of signal presence in the noisy observations. Signal absence in
the noisy observations is frequent, since speech signals contain large portions of silence. This absence of
signal implies its absence in the noisy spectral components as well. However, it is also possible that the
signal is present in the noisy observations, but appears with insignificant energy in some noisy spectral
components, which are randomly determined. This is a typical situation when the analyzed speech is of
voiced type, and the analysis is not synchronized with the pitch period. The above discussion suggests two
statistical models for speech absence in the noisy spectral components. In the first one, speech is assumed to
be either present or absent, with given probabilities, in all of the noisy spectral components. The reasoning
behind this model is that signal presence or absence should be the same in all of the noisy spectral
components, since the analysis is done on a finite interval. In the second model which represents the other
extreme, a statistically independent random appearance of the signal in the noisy spectral components is
assumed. As is implied by the above discussion, this model is more appropriate for voiced speech signals
when weak signal spectral components are considered as if they were absent.
4. Log Spectral Amplitude Estimator

This is the method used to calculate the a priori SNR of the signal. We now consider the estimation of
the a priori SNR of a spectral component by a decision-directed method. This estimator is found to be
very useful when it is combined with either the MMSE or the Wiener amplitude estimator [24].
An estimate of the a priori SNR is given as
2
( 1)

( ) (1 ) [ ( ) 1, 0], 0 1
( , 1)
k
k k
d
A n
n P n
k n
o
o o

= + s s

or
2

( ) ( ( 1), ( 1)) ( 1) (1 ) [ ( ) 1]
k k k k k
n G n n n P n o o = +
(5)
o is usually chosen to be 0.98 in order to get the best smoothing performance. The higher the o is, the
less musical noise, but the more distortion to the speech. Where

( 1)
k
A n is the amplitude estimator of the
kth signal spectral component in the (n - 1) th analysis frame, and P [.] is an operator which is defined by

0
[ ]
0
x ifx
P x
Otherwise
>
=

(6)
The proposed estimator for ( )
k
n is a decision-directed type estimator, since

( )
k
n is updated on
the basis of a previous amplitude estimate. Let ( )
k
n , ( )
k
A n , ( , )
d
k n and ( )
k
n denote the a priori
SNR [27], the amplitude, the noise variance, and the a posteriori SNR, respectively, of the corresponding
kth spectral component in the nth analysis frame. The derivation of the a priori SNR estimator is based here
on the definition of ( )
k
n , and its relation to the a posteriori SNR ( )
k
n , as given below:
2
{ ( )}
( )
( , )
k
k
d
E A n
n
k n

=

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or
( ) { ( ) 1}
k k
n E n = (7)
Several initial conditions were examined by simulations. We found that using
(0) (1 ) { (0) 1}
k k
P o o = + is appropriate, since it minimizes initial transition effects in the enhanced
speech. A priori SNR is defined as the ratio between the variances of the kth spectral components of the
speech and noise. It is denoted by ( , ) l k and given as
2
2
( , )
( , )
( , )
s
n
l k
l k
l k
o

o
=
(8)
A posterior SNR is defined as the ratio between the amplitude of the kth spectral components of the
noisy and variances of the kth spectral components of the noise. It is denoted by ( , ) l k and given as
2
2
( , )
( , )
( , )
n
R l k
l k
l k

o
=
(9)
According to the formulation of the estimation problem given above, we are looking for the estimator

k
A , which minimizes the following distortion measure:
2

E{(log log ) }
k k
A A (10)
Given the noisy observations {y(t), 0 < t < T}. This estimator is easily shown to be

exp{ [ln ( ), ]}
k k
A E A y t o t T = s s
(11)
and it is independent of the basis chosen for the log. Under the assumed statistical model, the expected value
of A
k
given {y (t), o < t < T} equals to the expected value of A
k
, given Y
k
only. Since this statement remains
true when A
k
is replaced by ln A
k
, the estimator equals,

exp{ [ln ]}
k k k
A E A Y = (12)
Note that the estimator results also if we choose to minimize the mean-square error of the log power
spectra given by,
2 2 2
{(log log ) }
k k
E A A
where
2
k
A denotes the estimator of
2
k
A , and use

k
A
= 2

k A
(13)
The evaluation of E[ln A
k
/ Y
k
] for the Gaussian model assumed here is conveniently done by utilizing
the moment generating function of ln A
k
given Y
k
. Let Z
k
= ln A
k
. Then The moment generating function
( )
k k
z Y
| of z
k
given Y
k
equals,
( )
k k
z Y
|
= {exp( ) }
k k
E Z Y
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= { }
k k
E A Y

(14)
E[ln A
k
/ Y
k
] is obtained from ( )
k k
z Y
| by

0
{ln } ( )
k k
k k z Y
d
E A Y
d

|

=
= (15)
Therefore, our task is now to calculate ( )
k k
z Y
| and then to obtain E[ln A
k
/ Y
k
] , ( )
k k
z Y
| is
given by
( )
k k
z Y
| =
1 2
0
0
2
0
0
exp( / ) (2 / )
exp( / ) (2 / )
k k k k k k k
k k k k k k k
a a I a v da
a a I a v da

}
}

Where
k
satisfies the following relation,

1 1 1
( ) ( )
k x d
k k
= +
(16)
And
k
v is defined by,

2
( )
; ;
1 ( ) ( )
k x k
k k k k
k d d
k R
v
k k



= = =
+
(17)

k
and
k
are interpreted as the a priori and a posteriori signal to- noise ratio (SNR), respectively. On
considering, we get the desired amplitude estimator,
1

exp
1 2
k
t
k
K k
k v
e
A dt R
t

=
`
+

)
}
(18)
It is useful to consider

k
A as being obtained from R
k
, by a multiplicative nonlinear gain function which
depends only on the a priori and the a posteriori SNR
k
and
k
, respectively. This gain function is defined
by

( , )
k
k k
k
A
G
R
=
(19)
The MMSE-STLSA estimator was implemented in the speech enhancement system, operating with the
decision- directed a priori SNR estimator. The residual noise obtained sounds a less uniform when the
MMSE STSA estimator is used. However, because of the lower residual noise level, this effect appears
insignificant. The reduction in the residual noise level obtained used is probably a result of the lower gain,
particularly in regions of low instantaneous SNR values. It is worthwhile noting that during this work we
also examined the STSA estimator which minimizes under the additional assumption that the signal is not
surely present in the noisy observation. This effect is reduced as the assumed probability of signal absence is
lowered; but then the amount of residual noise reduction gained by this estimator is also reduced. For the
above reasons we found it unworthy to incorporate the signal presence uncertainty in the log STSA estimator
[19].
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5. Analysis of Results and Discussions


(a)

(b)

(c)

(d)
Figure.3. (a), (b), (c), (d): Log-Spectrograms of denoised 5dBMozart, 4.75dBPiano, 10.76dBTIMIT and
5dBSpeech signal.

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(e)

(f)

(g)

(h)
Figure.4. (e), (f), (g), (h): Log-Spectrograms of denoised 2dBMozart, -5dBPiano, 20.63dBTIMIT and
8dBSpeech signal
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Figure.5.Comparison of Original, Noisy and Denoised Mozart Signal Graphs of MMSE-LSA.

0
5
10
15
20
25
Speech1 Mozart Speech2 TIMIT
S
S
N
R
V
a
l
u
e
s

Figure.6.Analysis of Results and Comparison of signals in form of charts
The STSA estimator was implemented in the speech enhancement system, operating with the decision-
directed a priori SNR estimator. It was examined by informal listening in enhancing speech degraded by
stationary uncorrelated additive white noise, with SNR values of 20.63, 10.76, 8, 5, 4.75, 2 and -5 dB
enhanced speech obtained and suffers much less residual noise, while no difference in the speech itself was
noticed [22]. The residual noise obtained with sounds a little less uniform than when the MMSE STSA
estimator is used. However, because of the lower residual noise level, this effect appears insignificant. The
reduction in the residual noise level obtained when a result of the lower gain, particularly in regions of low
instantaneous SNR values. Another interesting comparison is that of the STSA estimator with the MMSE
STSA takes into account signal presence uncertainty. We found that the enhanced speech obtained by both
estimators sounds very similar, with the exception that with the first estimator the residual noise sounds a
little less uniform. It is worthwhile noting that during this work we also examined the STSA estimator which
minimizes under the additional assumption that the signal is not surely present in the noisy observation.
While this estimator results in a further reduction of the residual noise in comparison with that it also
introduces an effect of low-pass filtering on the enhanced speech signal. This effect is reduced as the
assumed probability of signal absence is lowered; but then the amount of residual noise reduction gained by
this estimator is also reduced. For the above reasons we found it unworthy to incorporate the signal presence
uncertainty in the log STSA estimator.


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6. Summary and Conclusions

In this correspondence we derive a LSAE estimator which minimizes the mean-square error of the log-
spectra and examine it in enhancing noisy speech. We found that this estimator is superior to the MMSE
STSA estimator since it results in a much lower residual noise level without further affecting the speech
itself. In fact, the new estimator results in a very similar enhanced speech quality as that obtained with the
MMSE STSA estimator, which takes into account the signal presence uncertainty. So diagonal estimator is
very superior to MMSE STSA estimator.









Table 1. Comparison of Four Types of Noisy Signals with Different Noise Levels.
References
[1]. M. Bahoura and J. Rouat, Wavelet speech enhancement based on timescale adaptation, Speech
Commun., vol. 48, no. 12, pp. 16201637, Dec. 2006.
[2]. Y. Ephraim and D. Malah, Speech enhancement using a minimum mean-square error short-time
spectral amplitude estimator, IEEE Trans. Acoust., Speech, Signal Processing, vol. ASSP-32, pp.
1109- 1121, Dec. 1984.
[3]. R. M. Gray, A. Buzo, A. H. Gray, Jr., and Y. Matsuyama, Distortion measures for speech
processing, I EEE Trans.A coust, Speech, Signal processing, vol. ASSP-28, pp. 367-376, Aug. 1980.
[4]. Y. Ephraim and D. Malah, Speech enhancement using a minimum mean-square error log-spectral
amplitude estimator, Dep. Elec. Eng., Technion, Haifa, Israel, Tech.R ep. 488, Feb. 1984.
[5]. J. E. Porter and S. F. Boll, Optimal estimators for spectral restoration of noisy speech, in Proc.
IEEE Znt. Con$ Acoust, Speech, Signal Processing, Mar. 1984, pp. 18A.2.1-l8A.2.4.
[6]. W. A. Pearlman and R. M. Gray, Source coding of the discrete Fourier transform, ZEEE Trans.
Inform. Theory, vol. IT-24, pp. 683-692, Nov. 1978.
[7]. S. Gradshteyn and Z. M. Ryzhik, Table of Integrals, Series, and Products. New York: Academic,
1980.
[8]. IBM Application Program, System/360 Scientific Subroutine Package 360A-CM-O3X, Version 111,
pp. 368-369, 1968
[9]. J. S. Lim and A. V. Oppenheim, Enhancement and bandwidth compression of noisy speech, Proc.
IEEE, vol. 67, pp. 1586- 1604, Dec. 1979
[10]. J. L. Flanagan, Speech Analysis, Synthesis and Perception, 2
nd
ed. New York: Springer-Verlag,
1972,~. 210.
[11]. R. J. McAulay and M. L. Malpass, Speech enhancement using a soft-decision noise suppression
filter, IEEE Pans.Acoust.,Speech, Signal Processing, vol. ASSP-28, pp. 137-145, Apr. 1980.
[12]. D. Middelton, Introduction to Statistical Communication Theory. New York: McGraw-Hill, 1960, ch.
7, appendix 1.
[13]. W. A. Pearlman and R. M. Gray, Source coding of the discrete Fourier transform, IEEE Tbans.
Inform. Theory, vol. IT-24, pp.
Signal &SNR LASE
SNR SSNR
Mozart5dB
Mozart8dB
Mozart9.23dB
Mozart10dB
13.9771 14.4141
14.2619 14.4869
14.3259 14.5059
14.3564 14.5150
Piano4.75dB
Piano-5dB
13.7362 14.3474
13.8426 08.7953
TIMIT10dB
TIMIT10.76dB
TIMIT20.63dB
17.5989 14.2690
17.7268 18.4019
22.5146 18.2871
Speech5dB
Speech8dB
Speech10dB

05.8313 09.3837
17.0936 12.0435
16.9113 14.3012
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International Journal of Advances in Science and Technology,
Vol. 3, No.4, 2011



[14]. W. B. Davenport and W. L. Root, An Introduction to the Theory of Random Signals and Noise. New
York: McGraw-Hill, 1958, ch. 6.
[15]. J. M. Tribolet and R. E. Crochiere, Frequency domain coding of speech, IEEE naris. Acoust.,
Speech, Signal Processing, vol. ASSP-27, p. 522, Oct. 1979
[16]. R. Zelinski and P. Noll, Adaptive transform coding of speech signals, IEEE Pans. Acoust., Speech,
Signal Processing, vol. ASSP-25, p. 306, Aug. 1977
[17]. J. E. Porter and S. F. Boll, Optimal estimators for spectral restoration of noisy speech, in Roc. IEEE
Int. Conf Acoust., Speech, Signal Processing, Mar. 1984, pp. 18A.2.1-18A.2.4.
[18]. D. Middleton and R. Esposito, Simultaneous optimum detection and estimation of signals in noise,
IEEE Pans. Inform. Theory, vol. IT-14, pp. 434-444, May 1968.
[19]. Y. Ephraim and D. Malah, Speech enhancement under uncertainty of signal presence in the observed
signal, Dep. Elec. Eng., Technion, Haifa, Israel, EE pub. 543, July 1983.
[20]. T. T. Kadota, Optimal reception of binary Gaussian signals, Bell Syst. Tech. J., vol. 43, pp. 2767-
2810, Nov. 1964.
[21]. I. S. Gradshteyn and I. M. Ryzhik, Table of Integrals, Series, and Products. New York: Academic,
1980.
[22]. D. Middelton, The incoherent estimation of signal amplitude in normal noise backgrounds, in Time
Series Analysis, M. Rosenblatt, Ed. New York: Wiley, 1963, ch. 24.
[23]. Y. Ephraim and D. Malah, Speech enhancement using optimal nonlinear spectral amplitude
estimation, in Aoc. IEEE Int. Conf. Acoust., Speech, Signal Processing, Apr. 1983, pp. 1 11 8- 1121.
[24]. Speech enhancement using vector spectral subtraction amplitude estimation, in Proc. IEEE 13th
Conv. Elec. Electron.Eng. in Israel, Tel-Aviv, Mar. 1983.
[25]. R. E. Crochiere, A weighted overlap-add method of short-time Fourier analysis/synthesis, IEEE
Pans. Acoust., Speech, Signal Processing, vol. ASSP-28, pp. 99-102, Feb. 1980
[26]. T. Cai, Adaptive wavelet estimation: A block thresholding and oracle inequality approach, Ann.
Statist., vol. 27, pp. 898924, 1999.
[27]. I. Cohen, Speech enhancement using supergaussian speech models and noncausal a priori SNR
estimation, Speech Commun., vol. 47, no. 3, pp. 336350, Nov. 2005.
[28]. S. Ghael, A. Sayeed, and R. Baraniuk, Improved wavelet denoising via empirical Wiener filtering,
in Proc. SPIE, Math. Imag., San Diego, Jul. 1997.

Authors Profile

K. Sreedhar received the B.Tech. degree in Electronics and Communication Engineering from
JNTUH University, Hyderabad, India in 2005 and M.Tech degree in Communication Systems from
JNTUH University, Hyderabad, India in 2009. He attended the International Conference on
Technology and Innovation at Chennai. He also attended the National Conference at Coimbatore,
Tamilnadu, India on INNOVATIVE IN WIRELESS TECHNOLOGY. He is currently working as an
Assistant professor in Electronics and Communication Engineering department in VITS (N9)
Karimnagar, Andhra Pradesh, India. He has a Life Member ship in ISTE. His areas of interests are
Digital Signal Processing and Image Processing. He published three International papers.





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