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JP2011244306

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DESCRIPTION JP2011244306
PROBLEM TO BE SOLVED: To provide a technique of wavefront synthesis method that
reproduces a wavefront on the sound source side of a microphone array. A frequency domain
conversion unit 1 converts a sound signal collected by a microphone array into a frequency
domain signal by Fourier transform. The window function unit 2 multiplies the frequency domain
signal by the window function to generate a window function frequency domain signal. The filter
unit 3 applies a filter that simulates the wavefront at a position closer to the sound source than
the microphone array by ignoring the inhomogeneous wave to the frequency domain signal after
the window function. The time domain conversion unit 4 converts the signal after the application
of the filter into a time domain signal by inverse Fourier transform. [Selected figure] Figure 1
Sound field sound collecting and reproducing apparatus, method and program
[0001]
The present invention relates to a wave field synthesis technique in which sound signals are
collected by a microphone array installed in a certain sound field, and the sound field is
reproduced by a speaker array using the sound signals.
[0002]
Wavefront synthesis is a technology that virtually reproduces the sound field of a remote place
using a plurality of microphones and speakers.
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1
When performing the wavefront synthesis method, it is important to be able to freely control the
position of the wavefront for reproducing the sound field. This is because virtual images can be
given to the reproduced sound field and acoustic effects can be added.
[0003]
In the technique described in Non-Patent Document 1, it is possible to reproduce the wave front
A at a position farther from the sound source S than the microphone array M by using the sound
signal collected by the microphone array M shown in FIG. The
[0004]
Sascha Spors, Rudolf Rabenstein, and Jens Ahrens, “The Theory of Wave Field Synthesis
Revisited,” 124th Convention of the Audio Engineering Society Amsterdam, 2008 May 17-20
[0005]
However, in the technique described in Non-Patent Document 1, there is a problem that the
wavefront B (see FIG. 16) located closer to the sound source S than the microphone array M can
not be reproduced.
[0006]
In order to solve the above problems, the sound signal picked up by the microphone array is
converted into a frequency domain signal by Fourier transformation, and the frequency domain
signal is multiplied by the window function to generate a frequency domain signal after the
window function, and the microphone array A filter simulating the wavefront at a position closer
to the sound source is applied to the frequency domain signal after the window function, and the
signal after the application of the filter is converted into a time domain signal by inverse Fourier
transform.
[0007]
By applying the filter calculated by ignoring the inhomogeneous wave, it is possible to reproduce
the wave front B (see FIG. 16) on the sound source S side of the microphone array M.
[0008]
FIG. 1 is a functional block diagram of an example of a sound field collection and reproduction
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device according to a first embodiment.
FIG. 2 is a view for explaining an example of the arrangement of a microphone array and a
speaker array of the sound field collection and reproduction device according to the first
embodiment.
The functional block diagram of the example of the sound field sound collection reproducing |
regenerating apparatus of 2nd embodiment.
FIG. 2 is a view for explaining an example of the arrangement of a microphone array and a
speaker array of the sound field collection and reproduction device according to the first
embodiment.
The flowchart which shows the example of the sound field sound collection reproduction |
regeneration method.
The figure for demonstrating the example of the microphone of a dipole characteristic. The figure
for showing the conditions of simulation. The figure which shows the simulation result which
reproduced the sound field by reproducing the wave front of z = 0 by the conventional WFS. The
figure which shows the ideal sound field corresponding to FIG. The figure which shows the
simulation result which reproduced the sound field by reproducing the wave front of z = -0.6 m
by this invention. The figure which shows the ideal sound field corresponding to FIG. The figure
which shows the simulation result which reproduced the sound field by reproducing the wave
front of z = -2m by this invention. The figure which shows the ideal sound field corresponding to
FIG. The figure which shows the simulation result which reproduced the sound field with two
sound sources by this invention. The figure which shows the ideal sound field corresponding to
FIG. The figure for demonstrating the position of the wave front which it is going to reproduce.
[0009]
Hereinafter, an embodiment of the present invention will be described with reference to the
drawings.
[0010]
First Embodiment The sound field collection and reproduction apparatus and method according
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to the first embodiment are Nx × Ny monopole microphones disposed at the position z = z1 of
the first room shown in FIG. Using the sound signals collected by the two-dimensional
microphone arrays M1-1, M2-1,..., MNx-Ny, the wavefront B at the position z = z2 is calculated by
the method described below, and the second It reproduces with two-dimensional speaker arrays
S1-1, S2-1,..., SNx'-Ny 'composed of Nx'.times.Ny' monopole-shaped speakers arranged in the
room.
[0011]
Nx, Ny, Nx 'and Ny' are arbitrary integers.
Nx and Nx 'may be different.
Also, Ny and Ny 'may be different. That is, the number of microphones constituting the
microphone array and the number of speakers constituting the speaker array do not have to be
the same, and the microphones constituting the microphone array and the speakers constituting
the speaker array need to be in one-to-one correspondence. There is no.
[0012]
The positions of the microphones constituting the two-dimensional microphone arrays M1-1,
M2-1,..., MNx-Ny arranged at the position of z = z1 in the first room are represented by rs = (xi,
yj, z1) To The positions of the speakers constituting the two-dimensional speaker arrays S1-1, S21,..., SNx'-Ny 'disposed in the second room reproducing the wavefront B at z = z2 in the first room
Let rs' = (xn, yn, z2).
[0013]
The sound field sound collection and reproduction apparatus according to the first embodiment
includes, for example, a frequency domain conversion unit 1, a window function unit 2, a filter
unit 3, and a time domain conversion unit 4 as shown in FIG. Process.
[0014]
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The two-dimensional microphone arrays M1-1, M2-1,..., MNx-Ny arranged at the position z = z1
of the first room pick up the sound emitted by the sound source S of the first room Generate a
time domain sound signal.
The generated sound signal is sent to the frequency domain conversion unit 1. The sound signal
of time t collected in the microphone Mi-j of rs = (xi, yj, z1) is denoted as f (i, j, t).
[0015]
The frequency domain conversion unit 1 Fourier-transforms the sound signal f (i, j, t) collected by
the microphone arrays M1-1, M2-1, ..., MNx-Ny into the frequency domain signal F (i, j, t).
Convert to ω) (step S1). The generated frequency domain signal F (i, j, ω) is sent to the window
function unit 2. ω is a frequency. For example, the frequency domain signal F (i, j, ω) is
generated by short time Fourier transform. Of course, the frequency domain signal F (i, j, ω) may
be generated by another existing method.
[0016]
The window function unit 2 multiplies the frequency domain signal F (i, j, ω) by the window
function to generate a window function after frequency domain signal Fw (j, j, ω) (step S2). The
window function after frequency domain signal Fw (j, j, ω) is sent to the filter unit 3. As a
window function, a so-called Turkey window function w (i, j) defined by the following equation is
used, for example. Ntpr is a score to which a taper is applied, and is an integer of 1 or more and
Nx and Ny or less.
[0017]
[0018]
The filter unit 3 applies a filter simulating the wave front B at a position closer to the sound
source than the microphone array by ignoring the inhomogeneous wave to the frequency domain
signal Fw (j, j, ω) after the window function (step S3) ).
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If the signal after applying the filter is expressed as G (n, m, ω), the processing of the filter unit 3
is expressed as the following equation. k is a wave number, and when c is the speed of sound, k =
ω / c. e is the number of napiers and π is the circle ratio. The reason why the filter is expressed
by the following equation will be described later. G (n, m, ω) is sent to the time domain
conversion unit 4.
[0019]
[0020]
The time domain conversion unit 4 converts the signal G (n, m, ω) after the application of the
filter into a time domain signal g (n, m, t) by inverse Fourier transform (step S4).
The inverse Fourier transform may use an existing method. The time domain signals g (n, m, t)
are sent to the loudspeaker arrays S1-1, S2-1,..., SNx'-Ny '. The time domain signal g (n, m, t)
obtained for each frame by the inverse Fourier transform is appropriately shifted and linearly
summed to be a continuous time domain signal.
[0021]
The speaker arrays S1-1, S2-1,..., SNx'-Ny 'reproduce sound based on the time domain signal g (n,
m, t). More specifically, the speaker Sn-m reproduces sound based on the time domain signal g
(n, m, t) as n = 1,..., Nx ′, m = 1,. Thereby, the wave front B at the position z = z2 of the first room
is reproduced by the speaker arrays S1-1, S2-1,..., SNx'-Ny 'of the second room, and the sound of
the first room is reproduced. The place can be reproduced in the second room.
[0022]
As described above, by applying a filter that simulates the wavefront at a position closer to the
sound source than the microphone array by ignoring the inhomogeneous wave, it is possible to
reproduce the wavefront B on the sound source S side than the microphone array. . As shown in
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FIG. 2, when the sound source S is between the positions z = z1 and z = z2 in the first room, the
speaker arrays S1-1, S2-1,. It is possible to reproduce a sound field in which the sound source S
'is present in front of -Ny'.
[0023]
The size of the wavefront that can be reproduced is about the same as the area of the plane
collected by the microphone array.
[0024]
Dipole sound collection-monopole reproduction In the sound field sound collection and
reproduction apparatus according to the first embodiment, the microphone arrays M1-1, M2-1,
..., MNx-Ny are configured with microphones of dipole characteristics, and the speaker array S11, When S2-1,..., SNx'-Ny 'are configured by speakers having monopole characteristics, the filter
unit 3 may perform the processing represented by the following equation.
[0025]
[0026]
The microphone m1 of the dipole characteristic can be constituted by, for example, two
monopole microphone m2 as shown in FIG.
In this case, the difference between the sound signals collected by the two monopole
microphones m2 may be the sound signals collected by the dipole microphone m1.
In FIG. 6, the middle point of the two microphones m2 having monopole characteristics is the
position of the microphone m1 of the dipole characteristics, and the plane M of the microphone
arrays M1-1, M2-1,. Ru.
The straight line connecting the two monopole microphones m2 is orthogonal to the plane M of
the microphone arrays M1-1, M2-1, ..., MNx-Ny.
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[0027]
Monopole sound collection-dipole reproduction In the sound field sound collection and
reproduction apparatus according to the first embodiment, the microphone arrays M1-1, M2-1,...,
MNx-Ny are configured with microphones of monopole characteristics, and the speaker array S11. , S2-1,..., SNx'-Ny 'are configured by loudspeakers of dipole characteristics, the filter unit 3 may
perform the processing represented by the following equation.
[0028]
[0029]
The loudspeaker with the dipole characteristic can be configured by, for example, two monopole
loudspeakers, as with the microphone m1 with the dipole characteristic shown in FIG.
In this case, dipole characteristics can be realized by driving two monopole speaker speakers in
reverse phase.
The surface of the speaker array S1-1, S2-1, ..., SNx'-Ny 'is located at the middle point of the two
monopole speakers, and the straight line connecting the two monopole speakers is It is
orthogonal to the plane of the speaker arrays S1-1, S2-1, ..., SNx'-Ny '.
[0030]
Dipole Sound Collection-Dipole Reproduction In the sound field sound collection and
reproduction apparatus according to the first embodiment, the microphone arrays M1-1, M2-1,
..., MNx-Ny are configured with microphones of dipole characteristics, and the speaker arrays S11, S2 In the case where -1, ..., SNx '-Ny' are configured by loudspeakers of dipole characteristics,
the filter unit 3 may perform the process represented by the following equation.
[0031]
[0032]
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Second Embodiment While the sound field collection and reproduction apparatus and method of
the first embodiment use a two-dimensional microphone array and a two-dimensional speaker
array, the sound field collection and reproduction apparatus and method of the second
embodiment A one-dimensional microphone array and a two-dimensional speaker array are used.
As a result, the number of microphones, the number of speakers, and the number of channels can
be reduced, which makes the implementation relatively easy.
[0033]
The sound field collection and reproduction apparatus and method of the second embodiment is
a first-order apparatus including Nx monopole microphones disposed at the positions y = y1 and
z = z1 of the first room shown in FIG. 4. Using the sound signals collected by the original
microphone arrays M1-1, M2-1,..., MNx-1, the wavefront B at the position of y = y2, z = z2 is
calculated by the method described below, and the second It reproduces with one-dimensional
speaker arrays S1-1, S2-1,..., SNx'-1 'consisting of Nx' pieces of monopole characteristic speakers
arranged in the room.
[0034]
Nx and Nx 'are arbitrary integers.
Nx and Nx 'may be different.
That is, the number of microphones constituting the microphone array and the number of
speakers constituting the speaker array do not have to be the same, and the microphones
constituting the microphone array and the speakers constituting the speaker array need to be in
one-to-one correspondence. There is no.
[0035]
The positions of the microphones constituting the one-dimensional microphone arrays M1-1,
M2-1,..., MNx-1 arranged at the positions y = y1 and z = z1 of the first room are represented by rs
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= (xi, y1, Let's call it z1). Loudspeakers constituting one-dimensional loudspeaker arrays S1-1, S21,..., SNx'-1 arranged in the second room reproducing the wavefront B at y = y2, z = z2 of the first
room Let us denote the position of rs' = (xn, y2, z2).
[0036]
The sound field collection and reproduction device according to the second embodiment
includes, for example, a frequency domain conversion unit 1, a window function unit 2, a filter
unit 3, a time domain conversion unit 4, and a correction filter unit 5, as shown in FIG. The
process shown by the solid line and the broken line of
[0037]
The one-dimensional microphone arrays M1-1, M2-1,..., MNx-1 arranged at the positions y = y1
and z = z1 of the first room generate the sound emitted from the sound source S of the first
room. The sound is collected to generate a time domain sound signal.
The generated sound signal is sent to the frequency domain conversion unit 1. The sound signal
of time t collected in the microphone Mi-1 of rs = (xi, y1, z1) is denoted as f (i, t).
[0038]
The frequency domain conversion unit 1 transforms the sound signal f (i, t) collected by the
microphone arrays M1-1, M2-1,..., MNx-1 into a frequency domain signal F (i, ω) by Fourier
transformation. To do (step S1). The generated frequency domain signal F (i, ω) is sent to the
window function unit 2. ω is a frequency. For example, frequency domain signal F (i, ω) is
generated by short time Fourier transform. Of course, the frequency domain signal F (i, ω) may
be generated by another existing method.
[0039]
The window function unit 2 multiplies the frequency domain signal F (i, ω) by the window
function to generate a window function after frequency domain signal Fw (i, ω) (step S2). The
window function after frequency domain signal Fw (i, ω) is sent to the filter unit 3. As a window
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function, for example, a so-called Turkey key function wx (i) defined by the following equation is
used. Ntpr is a score to which a taper is applied, and is an integer of 1 or more and Nx or less.
[0040]
[0041]
The filter unit 3 applies a filter that simulates the wave front B at a position closer to the sound
source than the microphone array by ignoring the inhomogeneous wave to the frequency domain
signal Fw (i, ω) after the window function (step S3).
If the signal after the application of the filter is denoted as G (n, ω), the processing of the filter
unit 3 is expressed by the following equation. k is a wave number, and when c is the speed of
sound, k = ω / c. e is the number of napiers and π is the circle ratio. The reason why the filter is
expressed by the following equation will be described later. G (n, ω) is sent to the time domain
conversion unit 4.
[0042]
[0043]
Hn <(1)> (x) is a Hankel function of the first kind, and Hn <(2)> (x) is a Hankel function of the
second kind.
The Hankel function Hn <(1)> (x) of the first kind and the Hankel function Hn <(2)> (x) of the
second kind are Bessel functions Jn (x) of the first kind and Bessel functions Yn (x) of the second
kind. It is defined as follows using
[0044]
[0045]
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The correction filter unit 5 corrects G (n, ω) using the following correction filter in order to
correct an error caused by approximation with the primary array, and the corrected signal G ′
(n, ω) Are obtained (step S5).
[0046]
[0047]
In the case of a DC component with a fraction k = ω / c and ω = 0, the denominator jk of the
square root on the right side of the above equation is 0, so that the above correction filter can
not be applied.
Therefore, in the case of ω = 0, G ′ (n, ω) = G ′ (n, 0) = 0.
[0048]
The time domain conversion unit 4 converts the signal G ′ (n, ω) after applying the correction
filter into a time domain signal g (n, t) by inverse Fourier transform (step S4).
The inverse Fourier transform may use an existing method.
The time domain signals g (n, t) are sent to the loudspeaker arrays S1-1, S2-1,. The time domain
signal g (n, t) obtained for each frame by the inverse Fourier transform is appropriately shifted
and a linear sum is taken to be a continuous time domain signal.
[0049]
The speaker arrays S1-1, S2-1,..., SNx'-1 reproduce sound based on the time domain signal g (n,
t). More specifically, the speaker Sn-1 reproduces sound based on the time domain signal g (n, t)
as n = 1,. Thereby, the wave front B at the position of y = y2, z = z2 of the first room is
reproduced by the speaker arrays S1-1, S2-1,. The sound field of the room can be reproduced in
09-05-2019
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the second room.
[0050]
As described above, by applying a filter that simulates the wavefront at a position closer to the
sound source than the microphone array by ignoring the inhomogeneous wave, it is possible to
reproduce the wavefront B on the sound source S side than the microphone array. . As shown in
FIG. 4, when the sound source S is between the positions z = z1 and z = z2 in the first room, the
speaker arrays S1-1, S2-1,. A sound field in which the sound source S 'is on the front of -1 can be
reproduced.
[0051]
Dipole Sound Collection-Monopole Reproduction In the sound field sound collection /
reproduction device according to the second embodiment, the microphone arrays M1-1, M2-1,.
When S2-1,..., SNx'-1 are configured by speakers having monopole characteristics, the filter unit
3 may perform the processing represented by the following equation.
[0052]
[0053]
Monopole sound collection-dipole reproduction In the sound field sound collection and
reproduction apparatus according to the second embodiment, the microphone arrays M1-1, M21,... , S2-1,..., SNx'-1 are configured by loudspeakers of dipole characteristics, the filter unit 3 may
perform the processing represented by the following equation.
[0054]
[0055]
Dipole Sound Collection-Dipole Reproduction In the sound field sound collection / reproduction
device according to the first embodiment, the microphone arrays M1-1, M2-1,..., MNx-1 are
configured with microphones of dipole characteristics, and the speaker arrays S1-1, S2 In the
case where -1, ..., SNx'-1 are configured by loudspeakers of dipole characteristics, the filter unit 3
may perform the processing represented by the following equation.
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[0056]
[0057]
Theoretical Background The theoretical background of the present invention will be described.
Here, the reason why the processing of the filter unit 3 is the expression (2) when performing
dipole sound pickup-monopole reproduction in the first embodiment will be described as an
example.
In the following, ω may not be described to simplify the notation.
[0058]
Based on the Rayleigh equation, the sound pressure P (rs ') at the position rs' at z = z2 is
estimated using the sound pressure gradient ∂P (rs) / ∂z at the position rs at z = z1 in FIG. It
will be.
[0059]
[0060]
Here, κ <-1> is a kernel function.
[0061]
[0062]
kx and ky are the wave number in the x direction and the wave number in the y direction,
respectively.
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The kernel function <<-1> can be decomposed into the sum of the component κ1 <-1>
corresponding to the homogeneous wave and the component κ2 <-2> corresponding to the
inhomogeneous wave.
[0063]
[0064]
Here, in WFS, the component of the inhomogeneous wave does not contribute to the
reproduction of the sound field.
Therefore, the kernel function <<-1> can be approximated only by the component κ1 <-1>
corresponding to the homogeneous wave.
That is, the following equation is established.
[0065]
[0066]
By using this κ <-1>, the sound pressure P (rs ′) at the position rs ′ is expressed as follows.
[0067]
[0068]
When the speaker to be reproduced has a monopole characteristic, the sound field may be
reproduced by reproducing a signal obtained by doubling the sound pressure gradient ∂P (rs') /
∂z obtained by partially differentiating the sound pressure by z. it can.
[0069]
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[0070]
When the sound is picked up by a microphone with a dipole characteristic, the above expression
is discretely expressed in consideration of Fw (i, j, ω) = ∂P (rs, ω) / ∂z. (2).
[0071]
[0072]
Equations (1) and (3) can be derived from equations (1) and (3) in the same manner for the other
cases, although there is a difference whether the array is two-dimensional or one-dimensional.
Here, their derivation is omitted.
[0073]
[Simulation Result] As shown in FIG. 7, the sound signal of the sound field in the region of z <0 is
collected by the microphone array disposed at a distance of 4 cm in a straight line at the position
of z = 0, and the position of z = 0 A simulation was performed to reproduce the sound field using
a sound signal collected in the region of z> 0 with a speaker array arranged linearly at intervals
of 4 cm.
The total length of the microphone array and the speaker array is 3.84 m.
The sound source is a point sound source located at the position of (x, y, z) = (− 0.4 m, 0 m,
−1.0 m).
The sound source signal is a sine wave of 1 kHz.
[0074]
FIG. 8 is a simulation result of reproducing a sound field by reproducing a wavefront of z = 0 by
the conventional WFS, and FIG. 9 is an ideal sound field in this case.
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FIG. 10 is a simulation result of reproducing a sound field by reproducing a wavefront of z = -0.6
m according to the present invention, and FIG. 11 is an ideal sound field in this case.
FIG. 12 is a simulation result of reproducing a sound field by reproducing a wavefront of z = −2
m according to the present invention, and FIG. 13 is an ideal sound field in this case.
Comparing FIG. 8 with FIG. 12, it can be seen that in FIG. 12, the wavefront is synthesized as if it
were moved in the positive direction of the z axis.
Moreover, when FIG. 8 and FIG. 13 are compared, it turns out that the sound source is formed in
the front of a speaker array.
[0075]
FIG. 14 reproduces the wave front of z = -2.0 m when the sound source is at two positions of (0.4 m, 0 m, -1.0 m) and (0.6 m, 0 m, -1.2 m) It is a simulation result which reproduced a sound
field by the above, and is an ideal sound field in this case of FIG.
It can be understood from FIG. 14 that the sound field can be moved even when there are a
plurality of sound sources.
[0076]
[Modifications, Etc.] Each part constituting the sound field sound collecting and reproducing
apparatus may be provided in either the sound collecting apparatus arranged in the first room or
the reproduction apparatus arranged in the second room.
In other words, each processing of the window function unit 2 and the filter may be performed
by the sound collection device disposed in the first room, or may be performed by the
reproduction device disposed in the second room Good.
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The signal generated by the sound collection device is transmitted to the reproduction device.
[0077]
The positions of the first room and the second room are not limited to those shown in FIGS. 2
and 4.
The first room and the second room may be adjacent or separated from each other.
Also, the orientation of the first room and the second room may be any.
[0078]
The processing of the window function may be performed in the frequency domain or in the time
domain.
[0079]
The sound field sound collecting and reproducing apparatus can be realized by a computer.
In this case, the processing content of each part of this apparatus is described by a program.
And each part in this apparatus is implement | achieved on a computer by running this program
by computer.
[0080]
The program describing the processing content can be recorded in a computer readable
recording medium.
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Further, in this embodiment, these devices are configured by executing a predetermined program
on a computer, but at least a part of the processing contents may be realized as hardware.
[0081]
The present invention is not limited to the above-described embodiment, and various
modifications can be made without departing from the spirit of the present invention.
[0082]
Reference Signs List 1 frequency domain conversion unit 2 window function unit 3 filter unit 4
time domain conversion unit 5 correction filter unit
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