Electronics and Communications in Japan, Vol. 100, No.
2, 2017
Translated from Denki Gakkai Ronbunshi, Vol. 136-C, No. 2, February 2016, pp. 108115
An Adaptive Notch Gain Using an Inverse Notch Filter and a Linear Prediction
Filter
YOUHEI NAKAMURA, ARATA KAWAMURA, and YOUJI IIGUNI
Graduate School of Engineering Science, Osaka University, Japan
SUMMARY
This paper proposes an adaptive notch filter that au-
tomatically adjusts the notch frequency and notch gain,
where they are the null frequency and the depth of the
null, respectively. The proposed notch filter consists of an
inverse notch filter and a linear prediction filter, and gives
an appropriate filter gain which can eliminate a sinusoidal
noise and extract only a wide-band signal even if they
share the same frequency. We compare the capability of the
proposed notch filter with traditional methods. Simulation
results show that the proposed system improves the SNR
in comparison to the conventional methods. C 2017 Wiley
Periodicals, Inc. Electron Comm Jpn, 100(2): 5867, 2017;
Published online in Wiley Online Library (wileyonlineli-
brary.com). DOI 10.1002/ecj.11935
Keywords: notch gain; notch filter; linear prediction
filter; inverse notch filter.
1. Introduction
In recent years, in the fields of medical science,
communications, phonetics, and imaging, many studies on
removal of single or multiple sinusoidal noise have been
reported [15]. Traditional linear prediction filter can be
used to achieve this purpose [6]. A linear prediction filter is
the filter to estimate current signal by linearly coupling the
past input signals. The correlation between the sinusoidal
noise and the past signal is 1, and hence as long as a linear
prediction filter has sufficiently large degrees, it can predict
a correct signal completely [7]. Therefore, if prediction
error is treated as output, a signal that only sinusoidal noise
is removed can be obtained. However, multiple filter co-
efficients are needed to realize steep band-stop function,
and hence the increase of arithmetic operations for coef-
ficient update is inevitable. A typical method to remove
58
sinusoidal noise with less degree is a notch filter [1]. The
notch filter is a narrow band stop filter that removes only a
specific frequency (named as notch frequency). Especially,
if the frequency of mixed sinusoidal noise is unknown, an
adaptive notch filter is effective. As regards the studies on
adaptive notch filter, a frequency estimation algorithm and
its fast convergence have been reported [812]. Many kinds
of frequency estimation algorithms have already been re-
ported. For example, Nishimura and colleagues conducted
the update of filter coefficients using stochastic gradient
descent [1]. On the other hand, although filter configuration
is the same as the previous one, Okello and colleagues
conducted the update of filter coefficient using simplified
stochastic gradient descent [13]. The update methods of
these adaptive algorithms are different from each other, but
it is demonstrated that the expected value of filter coeffi-
cient converges to true value in each method [1, 14]. In
general, there is a trade-off between convergence speed and
estimation accuracy, but it is reported that both convergence
speed and estimation accuracy can be improved simulta-
neously by connecting two notch filters with distinctive
characteristics in parallel [2, 15, 16]. Also, even if multi-
ple sinusoidal noises exist, the problem can be overcome
by connecting notch filters in cascade, but simple cascade
connection causes frequency estimation error. On the other
hand, it is also reported that frequency estimation error can
be reduced by using output error at the final stage to update
filter coefficient at each stage [14]. However, if the desired
signal and the sinusoidal noise share the same frequency,
the gain at notch frequency (named as notch gain) is zero,
and hence a part of desired signal is deteriorated even if
frequency estimation is correct. To prevent the degradation
of desired signal, it is necessary to provide suitable notch
gain. Bae and colleagues proposed a notch filter to adjust
notch gain at the zero point of the filter [17]. However,
the adaptation of gain according to intensity of sinusoidal
noise was not investigated, because it is difficult to solve it
analytically. Sugiura and colleagues proposed the method
C 2017 Wiley Periodicals, Inc.
to adjust notch gains adaptively using the comb filter [18].
The comb filter is the filter with gains at a regular interval,
and the method proposed by Sugiura and colleagues can
adjust each notch gain independently. Thus, single notch
filter can be achieved by adjusting the notch gains other than
1 to 1. It should be noted that this method has a problem
and that it is difficult to adapt to unknown frequency be-
cause notch frequency is determined by delay device inside
filter.
To solve the problems described above, in this paper,
we propose a notch filter with fewer arithmetic operations
to adaptively adjust notch frequency and notch gain by
using input signal, which is the sum of unknown sinusoidal
noise and wide-band desired signal. It is assumed that the
desired signal is not correlated with sinusoidal noise, and
autocorrelation function of the desired signal turns to zero
within a short time difference. For convenience, in this
paper, it is discussed with white signal, in which its autocor-
relation function is 0 for any time difference except 0, as the
desired signal. In the proposed system, the sinusoidal noise
and a part of the desired signal, of which frequency is the
same as that of the sinusoidal noise, are removed first using
conventional notch filter with zero notch gain [13]. Then,
only sinusoidal noise is extracted from the removed signal,
and it is subtracted from the input signal to achieve adaptive
notch gain. Here, the signal removed by notch filter can be
obtained directly as an output of inverse notch filter [19].
The inverse notch filter is a narrow-band band-pass filter
with inverse characteristics of notch filter. It is assumed that
correlation of the desired signal included in inverse notch
filter is shorter than that of sinusoidal noise, and only the
sinusoidal noise is extracted using linear prediction filter
[20]. In this work, for the autocorrelation function r x x () =
E[x(n)x(n )] of the signal x(n), when r x x () = 0, || >
T , the correlation length of x(n) is expressed as T. Here, the
desired signal passing through the inverse notch filter has
a correlation length equivalent to its impulse response [6].
It is known that impulse response of inverse notch filter
approaches to zero with tens to hundreds of samples [19].
Accordingly, if forward prediction of hundreds of steps
is conducted using linear prediction filter, only sinusoidal
noise can be extracted. Meanwhile, the signal to be pre-
dicted by linear prediction filter is already a narrow-band
signal, and therefore linear prediction filter itself does not
need to realize steep frequency characteristics. Therefore,
the degree of the linear prediction filter used in the pro-
posed system is lower than that of conventional one, and
it is possible to realize it. Sinusoidal noise is obtained as
a prediction value of linear prediction filter, so that if it is
subtracted from input signal, an adaptive notch gain can be
realized.
This paper is organized as follows. In Section 2, the
way to extract sinusoidal noise from output of inverse notch
filter is described, and then the entire proposed system is
derived. In Section 3, the effectiveness of the proposed
59
Fig. 1. Basic structure of notch filter with adaptive gain.
system is verified by simulation. Finally, in Section 4, we
conclude this paper.
2. Realization of Adaptive Notch Gain Using Inverse
Notch Filter and Linear Prediction Filter
The input signal x(n) at time n is the sum of the
desired signal w(n) and the sinusoidal noise s(n), and it is
expressed as follows:
x (n) = w (n) + s (n) , (1)
where it is assumed that the desired signal satisfies the
following condition for rww () = E[w(n)w(n - )].
{ 2
, =0
rww () = , (2)
0 otherwise
where 2 is a constant. In this paper, it is supposed that
the desired signal is a Gaussian white signal (average: 0,
variance: 2 ). Furthermore, s(n) is the sum of sine curves
as follows:
K
( )
s (n) = pk cos k n + k , (3)
k=1
where K is the number of sine curves, k , pk , and k are the
frequency, amplitude, and initial phase of k-th sine curve,
respectively. It should be noted that k is the probability
signal and is distributed uniformly between and . s(n)
is not correlated with w(n). In this paper, the number of
sinusoidal noise is determined as K = 1 to simplify the
discussion. If K 2, it can be realized by connecting the
proposed system in cascade, as the same way as conven-
tional one [13, 15].
2.1 Wiener gain to remove sinusoidal noise
The filter configuration as shown in Fig. 1 is con-
sidered as the way to remove the sinusoidal noise with-
out degrading the desired signal. In this figure, BPF is a
narrow-band band-pass filter to extract the component near
frequency 1 . The frequency characteristic of BPF is B(),
and it satisfies the conditions as follows:
|B ()| = 1, (4)
( )
B 1 = 2m (m = 0, 1, 2, ) . (5)
yB (n) is the signal passing through the band-pass filter, G is
the gain, and y(n) is the output signal of the entire filter. In
the entire filter, the notch filter is configured equivalently.
If m-th impulse response of BPF is denoted by hm , the
following equation can be obtained in steady state:



y B (n) = h m w (n m) + h m s (n m)
m=0 m=0

= h m w (n m)
m=0
Fig. 2. Structure of inverse notch filter.
| ( )| ( ( ))
+ | B 1 | p1 cos 1 n + 1 + B 1
| |

= h m w (n m) + s (n), (6)
m=0
where Eq. (3) (K = 1) as well as Eqs. (4) and (5) are utilized.
G is solved to minimize the root mean squared error of s(n)
and GyB (n). The evaluation function is as follows:
[( )2 ]
J = E s (n) G y B (n) . (7)
G to minimize Eq. (7) is called Wiener gain, which
is:
p12
G opt = . (8)
2 2 2
m=0 h m + p12
In the system as shown in Fig. 1, we adaptively realize
Wiener gain shown in Eq. (8) to investigate the way to
construct notch filter equivalently.

2.2 Inverse notch filter and its impulse response

In the previous section, we demonstrated the gain

targeted in this study. In this section, the notch filter, which
can be used as BPF in Fig. 1, is explained.
The transfer function of the inverse notch filter can
be expressed as follows [19]:
Fig. 3. Amplitude and phase responses of the inverse
1r 1 z 2 notch filter. [Color figure can be viewed in the online
I (z) = , (9)
2 1 + z 2 + r z 2 issue, which is available at wileyonlinelibrary.com.]
where r (0 < r < 1) is a constant to determine the pass-band
width of the inverse notch filter, and is the filter coefficient
u (n) = x (n) u (n 1) r u (n 2) . (12)
to determine notch frequency (N ) so that the gain is 1, and
it is expressed as follows: Figure 3 shows the frequency amplitude character-
istics and the phase response property of the inverse notch
= (1 + r ) cos N . (10)
filter for various r (constant) at = 0. Figure 3 demonstrates
Thus, if N = 1 , 1 component of the input signal that this filter functions as band-pass filter and that the
is included in the output of inverse notch filter. Figure 2 amplitude is 1 at the notch frequency, and the amplitude
shows the block diagram of inverse notch filter, where x(n) approaches 0 as it goes away from the notch frequency. The

is the input signal, u(n) is the state variable, and x(n) is the more the constant r approaches 1, the steeper the amplitude

output signal of the inverse notch signal. x(n) and u(n) are, response is. Furthermore, the phase response is 0 at the
respectively, expressed as follows: notch frequency. Accordingly, the inverse notch filter satis-
1r fies the feature of Eqs. (4) and (5). Therefore, it is confirmed
x (n) = (u (n) u (n 2)) , (11) that it can be utilized as BPF in Fig. 1.
2

60
 Table 1. Convergence length for sum of squared impulse
responses

N
r 0.2 0.25 0.5 0.75 0.8
0.8 35 35 34 35 35
0.9 73 72 72 72 73
0.98 376 376 376 376 376

indicates that the sum of squared impulse response con-

Fig. 4. Convergence property for sum of squared impulse verges to the calculated value given in Eq. (21). It is further
responses. [Color figure can be viewed in the online issue, confirmed that the convergence speed of J(n) is slower as r
which is available at wileyonlinelibrary.com.] approaches 1. Here, the convergence ratio can be evaluated
by the ratio of Eqs. (18) and (21) (i.e., J(n)/J()). J(n) is
a monotonically increasing function. If the threshold (Th)

Here, x(n) has transient state, and its length is equal
is about 1, the minimum n satisfying the following Eq.
to the impulse response of inverse notch filter. Thus, we
(22) can be treated as the length until convergence approx-
first verify the impulse response of the inverse notch filter.
imately:
As reported in Ref. [19], the impulse response hn of inverse
notch filter at n = 0, 1 is: J (n) rn
=1 {q c (n)} T h. (22)
1r J () p2
h0 = , (13)
2 For example, Table 1 shows n which satisfies Eq. (22)
when Th = 0.9995 and N as well as r that are varied.
1r
h 1 = . (14) From the result, it is found that n almost converges with
2
approximately 35, 72, and 376 samples at r = 0.8, 0.9,
The impulse response (n 2) is and 0.98, respectively. That is to say, it is considered that
1 r (n1)2 these numbers of samples represent the length of impulse
hn = r {r sin (n + ) sin (n )} , (15)
p response of inverse notch filter against r. Based on Table 1,
where it is also confirmed that the length of the impulse response
does not depend on notch frequency, but it is determined by
p = 4r 2 , (16) r. Because the convergence of the sum of squared impulse
( p ) response of inverse notch filter J(n) is given by Eq. (21), the
= arctan . (17) desired gain in this paper is as follows:

p is a positive real number. That is, it is assumed that 2 < p12
4r. This condition is satisfied when r  1 [19]. The sum G opt = , (23)
(1 r ) 2 + p12
of squares up to n-th impulse response (denoted by J(n)
2 1r
hereafter) can be expressed as follows: where m=0 h m =
[ ] 2
1r rn
J (n) = 1 {q c (n)} , (18)
2 p2
2.3 Autocorrelation function of output of
where inverse filter
q = (1 + r )2 2 , (19)
In this section, we describe the correlation between
c (n) = (1 r ) {cos (2n) r cos (2 (n + 1))} . (20) the autocorrelation of output of inverse notch filter with the
impulse response when white signal is input. This analysis
Because r < 1, the convergence of the sum of squared is necessary to construct the system to obtain adaptive notch
impulse response can be expressed as follows based on Eq. gain.
(18): The output of inverse notch filter is expressed as fol-
1r lows:
lim J (n) = . (21)
n 2
x (n) = w N (n) + s N (n) , (24)
This equation reveals that the convergence of J(n)
does not depend on notch frequency, but just relies on r. where wN (n) is the signal derived from the white sig-
Here, Fig. 4 shows the case of J(n) at = 0. This figure nal, and sN (n) is the signal derived from the sinusoidal

61
noise. Here, the autocorrelation function wN (n) is defined as
follows:
[ ]
rw N () = E w N (n) w N (n ) . (25)
If the m-th impulse response of inverse notch filter is
denoted by hm , it is:


w N (n) = h m w (n m), (26)
m=0
and hence the following equation is obtained:
[ ]
rw N () = h m h l E w N (n m) w N (n l + )
m l .
(27)
= 2 h m h m
m
Therefore, if integer M satisfies hm  0 (m M), then
rw N () = 0(|| M). Here, Fig. 5 shows rw N () for r =
0.8, 0.9, and 0.98 at = 0, respectively. These results are
the averages of 100 trials. These figures demonstrate that
the autocorrelation lengths of output of inverse notch filter
for white signal are approximately 50, 90, and 400 samples
at r = 0.8, 0.9, and 0.98, respectively. This result is almost
consistent with that shown in Table 1.

2.4 System to obtain adaptive notch gain

In the system shown in Fig. 1, the method to de-

sign G, in which the optimal value is given by Eq. (8), is
important. In the proposed system, the way to automati-
cally design G is investigated using the linear prediction
filter.
In the previous section, we described the length of
autocorrelation for white signal given by the inverse notch
filter. From this result, when the length of impulse re-
sponse of the inverse notch filter is denoted by M, only
sinusoidal noise can be estimated from x(n) by using linear
Fig. 5. Autocorrelation of w N (n). [Color figure can be
prediction filter provided with delay D ( M). The desired
viewed in the online issue, which is available at
signal can be obtained by subtracting the estimated sinu-
wileyonlinelibrary.com.]
soidal noise from the input signal. As an overall system,
notch filter with adaptive gain is constructed. Thus, the
linear prediction filter is responsible to realize the adaptive In this paper, as the update algorithm for gi (n), the
gain. Figure 6 shows the block diagram of the proposed representative normalized least mean square (NLMS) al-
system, where I(z) represents the inverse notch filter. The gorithm [21] is used as follows:
symbol y(n) is a signal subtracting output x(n) of I(z)
e (n) x (n i D)
from x(n), s (n) is the predicted value of linear prediction gi (n + 1) = gi (n) + L , (29)
N 1 2
filter, and e(n) is the prediction error. y(n) is the output l=0 x (n l D)
signal of the entire system. Here, P(z) is the transfer func- where L is the step size. The inverse notch filter and the
tion of the linear prediction filter, and it is expressed as notch filter have one-to-one relationship, and hence the
follows: frequency estimation algorithm for notch filter can be used

N 1 without modification. As a frequency estimation algorithm
P (z) = gi (n) z iD , (28) of notch filter, the stochastic gradient descent is known [1],
i=0 but this method needs square root calculation every time
where gi (n) is the filter coefficient of linear prediction filter, when filter coefficient is updated. On the other hand, the
and N is the degree of filter. more simplified stochastic gradient descent [13, 14] does

62

Fig. 6. Structure of proposed system.
Fig. 7. Structure of the lower part of the proposed
system.
not need such calculation, and moreover its convergence is
ensured. Thus, in this paper, from the viewpoint of arith-
metic operations, the filter coefficient of inverse notch filter
((n)) is updated using the simplified stochastic gradient
descent [13]. The update equation can be expressed as fol-
lows:
(n + 1) (n) I u (n 1) y (n) , (30)
where L is the step size. In the proposed system, frequency
provides both the notch frequency and the notch gain adap-
tively for unknown sinusoidal noise.
2.5 Correlation between proposed system and
conventional linear prediction filter
In this section, we investigate the difference between
the proposed system and the conventional linear prediction
filter. Figure 7 shows the lower path of the proposed sys-
tem shown in Fig. 6. The part boxed by dash line works
as a linear prediction filter. A conventional linear predic-
tion filter can extract or remove sinusoidal noise included
in white signal by giving proper delay and sufficient de-
grees. In such application, linear prediction filter is often
called adaptive line enhancer [20, 22]. The convergence
value of the linear prediction filter against the input sig-
nal expressed by Eq. (1) at K = 1 can be expressed as
follows [22]:
2a [ ]
lim g (n) = cos 1 (i + D) , (31)
n i N ing SNRs were used:
where
N
a= 2N , (32)
1+ 2
p12
= . (33)
22
As a fixed N, a approaches 0 when 2 increases, and
hence gi (n) also approaches 0. This indicates that sine curve
is embedded in white signal and the estimation becomes
difficult. If N , it results in a 1. This is the same
effect as 2 0. Therefore, if the linear prediction filter
with higher degrees is used, the sinusoidal noise can be
extracted at a high accuracy. It is natural that a does not
approach 1 if the linear prediction filter with low degrees
is applied. Therefore, the conventional prediction filter is
required to approach a to 1 at high degrees.
On the other hand, the proposed system can adjust 2
in Eq. (33) to 1r
2
2 using the inverse notch filter I(z). Thus,
with r  1, the linear prediction filter at poststage (P(z)) can
realize a  1 even with low degrees. Here, the convergence
values of the filter coefficients of conventional linear pre-
diction filter and the one used in the proposed system are
compared by simulation. The degree of linear prediction
filter (N) was 22 and D = 400, and the input signal was
generated by adding the sinusoidal noise to Gaussian white
signal (average and variance are 0 and 1, respectively) so
that = 0.3. The frequency of sinusoidal noise was 1 =
0.25. Meanwhile, the NLMS algorithm was used to update
filter coefficient, and its stepping size was 0.001. The con-
dition r = 0.98 was used for inverse notch filter. Figure 8
shows the filter coefficient of linear prediction filter after
convergence, where the solid lines are coefficients obtained
by simulation, and the dotted lines are the values calculated
by Eq. (31) when a = 1 (2 = 0). Figure 8(a) indicates
that the coefficient of conventional linear prediction filter
approaches 0, which is difficult for the accurate prediction
of sine wave. Here, according to Eq. (32), a = 0.78. On
the other hand, Fig. 8(b) indicates that the coefficient of the
proposed system is almost the same as that calculated when
2 = 0. In this case, a = 0.99. Therefore, it is confirmed
that 2 is reduced by inverse notch filter, and as a result the
system is robust against the white signal influence even with
low degrees. That is to say, the proposed system realizes the
reduction of arithmetic operations with lower filter degrees
of linear prediction filter due to the effect of I(z).
3. Performance Evaluation by Simulation
The effectiveness of the proposed system was verified
by simulation. For the performance evaluation, the follow-
( [ ])
E w 2 (n)
Input SNR = 10log10 [ ] [dB] , (34)
E s 2 (n)
63
 Table 2. Number of calculations of LPF and the
proposed system

Addition Multiplication
LPF 2Nconv 1 3Nconv
Prop. 2N+3 3N+5

Table 3. Results of Output SNR (unit: dB) for noise

reduction

Input ANF LPF Prop. Optimal gain

0.0 20.1 11.0 21.6 21.7
5.0 20.7 12.1 21.7 21.9
10.0 20.8 13.6 22.1 22.4
15.0 20.9 16.1 22.4 23.2

Table 2 shows the respective number of multiplication and

Fig. 8. Comparison of the filter coefficient of the linear
prediction filter. [Color figure can be viewed in the online
issue, which is available at wileyonlinelibrary.com.]
( [ ] )
E w 2 (n)
Output SNR = 10log10 [ ] [dB] , (35)
E (w (n) y (n))2
where Output SNR becomes the maximum () only
when output y(n) matches with the desired signal w(n) at
wave level. It should be noted that expected value cannot
be calculated actually, and hence it is replaced by the time
mean of signals of 1000 samples after convergence. The
input signal was generated by adding sinusoidal noise to
Gaussian white signal (average and variance is 0 and 1,
respectively) so that Input SNR is 0, 5, 10, and 15 dB,
where the frequency of sinusoidal noise (1 ) is 0.5. For
the inverse notch filter of the proposed system, r = 0.98
and step size I = 0.001 were used. As shown in Fig. 4
in Section 2.2, the impulse response of the inverse notch
filter converges with about 400 samples at r = 0.98, and
therefore the delay of linear prediction filter (D) was set as
400, and N = 20 as well as L = 0.001. The conventional
linear prediction filter [6] and the adaptive notch filter [13]
were used for comparison. The filter degree (Nconv ), the
delay value (D), and the step size (L ) of the conventional
linear prediction filter were 22, 1, and 0.001, respectively.
Here, the degree of filter was selected so that the number
of multiplications and additions, which is needed to obtain
the final output, is equivalent to that of the proposed system.
addition, where LPF and Prop mean a linear prediction filter
and the proposed system, respectively. The conventional
adaptive notch filter was designed so that r = 0.98 and I =
0.001.
Table 3 shows the result of Output SNR, where each
result is the mean of 100 trials, and ANF in the table
represents the adaptive notch filter. For reference, the op-
timal gain is also listed, which is obtained by designing
the conventional adaptive filter at reduction frequency (1 )
and providing the optimal notch gain (1-Gopt ). Specifically,
BPF in Fig. 1 was designed by inverse notch filter with
N = 1 , which is equivalent to the one with G = Gopt .
The result in Table 3 demonstrates that SNR of the proposed
system is improved well, compared to linear prediction
filter and adaptive notch filter. Especially when the input
is 0 dB, the Output SNR of the proposed system was im-
proved by 1.5 dB and 10.6 dB, compared to adaptive notch
filter and linear prediction filter, respectively. Also, when
compared with the theoretical value, it is confirmed that
the performance of the proposed system almost reaches the
theoretical value.
Then, the gains for 1 at various Input SNRs were
investigated to evaluate the adaptive notch gain of the pro-
posed system. The gains for 1 obtained by the conven-
tional linear prediction filter and the adaptive notch filter
were further compared with those of the proposed system.
Figure 9 shows the results, where each of them is a mean
of 100 trials, and the optimal value in Eq. (23) is also
presented. In this figure, the horizontal and vertical axes
refer to the Input SNR and the intensity of gain for 1 ,
respectively. The gain of the conventional linear prediction
filter is larger than that of the optimal value. In this case,
the sinusoidal noise cannot be removed completely, which
remains in the output signal. It is found that the gain for 1
remains almost zero against the change of the Input SNR,

64
Fig. 9. Notch gain in each input SNR. [Color figure can
be viewed in the online issue, which is available at
wileyonlinelibrary.com.]

because the notch gain of conventional adaptive notch filter

is fixed to zero. In this case, a part of the desired signal
is lost. However, the gain slightly increases at high SNR.
It is considered because the power of sinusoidal noise is
low and the error of frequency estimation formed. On the
other hand, the gain of the proposed system is more close
to the theoretical value, compared to the conventional linear
prediction filter and the adaptive notch filter. That is, the de-
terioration of desired signal is limited. However, sinusoidal
noise is overwhelmed in white signal at high Input SNR,
and thus it is considered that the accuracy of frequency
estimation is deteriorated and the difference between the
gain for 1 and the optimal one increases. The larger the
SNR is, the more significant the difference between the gain
of the proposed system and the optimal one becomes. This
tendency is consistent with the Output SNRs in Table 3.
Finally, the ultimate frequency amplitude responses
of the linear prediction filter, the adaptive notch filter, and
the proposed system at Input SNR = 15 dB are compared,
where each value is also the mean of 100 trials. First, the
frequency amplitude responses of the conventional linear
prediction filter are shown in the upper inset of Fig. 10, Fig. 10. Amplitude response of the conventional
where the solid and dotted lines are the frequency ampli- methods and proposed filter at Input SNR = 15 dB. [Color
tude response obtained by simulation and the theoretical figure can be viewed in the online issue, which is available
response of the notch filter at r = 0.98 with the optimal at wileyonlinelibrary.com.]
gain, respectively. This figure demonstrates that the gain
for 1 of the linear prediction filter is much larger than amplitude responses of the proposed system are shown in
the theoretical one. It is also confirmed that narrow-band the lower inset of Fig. 10. This figure demonstrates that the
amplitude response is not realized and multiple ripples proposed system realizes narrow-band amplitude response,
are generated. Then, the frequency amplitude responses of and it approaches the gain for 1 to the theoretical value.
the adaptive notch filter are shown in the middle inset of However, a slight error between the notch frequency and 1
Fig. 10. This figure demonstrates that narrow-band ampli- is found as the case of the adaptive notch filter in the middle
tude response is realized, while the gain for 1 is lower inset of Fig. 10, where the error of coefficient was 1.4
than the theoretical value and it is almost 0. It should be 103 . It was confirmed from these results that the proposed
noted that the notch frequency was slightly different from system is effective to remove sinusoidal noise compared to
1 , and the difference to the real coefficient value was 1.6 the conventional system when sinusoidal noise is relatively
103 . This indicates that the error arises to the frequency low, although there forms a slight error of frequency esti-
estimation of sinusoidal noise. In the end, the frequency mation slightly.

65
 4. Conclusion
In this paper, we proposed the configuration of the
notch filter to adaptively adjust a notch gain, which was
conventionally given as a fixed value. The proposed system
consists of an inverse notch filter and a linear prediction
filter. The proposed system conducts a linear prediction
against the output of inverse notch filter, and the predicted
value is subtracted from the input signal to achieve the
adaptive notch gain. The result of simulation shows that
the notch gain of the proposed system is near the opti-
mal gain. The results confirm that the Input SNR of the
proposed system is, respectively, improved by 1.5 dB and
10.6 dB compared to the conventional adaptive notch filter
and the linear prediction filter with equivalent arithmetic
operations, which support the effectiveness of the proposed
system.
In the future, we will conduct the performance evalu-
ation when other adaptive algorithm (e.g., stochastic gradi-
ent descent) [1] is combined with the proposed system, and
investigate the combination of the proposed system with
various configurations of various notch filters.
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 AUTHORS (from left to right)

Youhei Nakamura (nonmember) graduated from the Department of Systems Science, School of Engineering Science,
Osaka University in March 2013. Nakamura completed the Masters program at Graduate School of Engineering Science,
Osaka University in March 2015. During his study, he was engaged in studies related to digital signal processing.

Arata Kawamura (nonmember) graduated from the Department of Electrical and Electronic Engineering, School of
Engineering, Tottori University in March 1995. Kawamura completed the Masters program at Graduate School of Engineering,
Tottori University in March 2001. He became Research Associate at Osaka University in 2003. After becoming an Assistant
Professor at the same university in 2007, he became Associate Professor at the same university in 2012. He is engaged in studies
related to signal processing. He is a Doctor of Engineering.

Youji Iiguni (nonmember) graduated from the Faculty of Engineering, Kyoto University in March 1982. Iiguni completed
the Masters program at the same university in 1984, and became Research Associate in the same year. After becoming an
Assistant Professor at Osaka University in 1995, he became Professor at the same university in 2003. He is engaged in studies
related to system analysis. He is a Doctor of Engineering.

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