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JPH01236899

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DESCRIPTION JPH01236899
[0001]
BACKGROUND OF THE INVENTION 1. Field of the Invention The present invention relates to a
sound receiving device for blocking an interfering sound incident from the outside and detecting
only a desired sound. The present invention is a conference in which it is desirable to be able to
perform speech as a general input microphone of a speech recognition apparatus in which the
necessity of suppressing and operating an acoustic signal other than the target voice as much as
possible is also possible. Especially as microphones for various music or dramas where it is
desired to separate and pick the sounds of each part without interfering with each other, as
microphones for public telephones or general public address systems, or as microphones for
various music or dramas where it is desired to It is suitable for use as an extremely high
performance super close-talking microphone that is desired in high environments. [Summary]
The present invention provides a sound receiving apparatus including means for removing
extraneous noise components contained in an acoustic signal detected by a central microphone,
wherein a boundary surrounding a plurality of sub microphones including two sound receiving
elements is surrounded by the central microphone. By arranging the signals on the surface and
processing the outputs of the sub microphones respectively to obtain noise components, noise
coming from the outside of the boundary surface is removed and a super close talk microphone
is realized. 2. Description of the Related Art Techniques for blocking external noise outside the
desired volume to improve the signal-to-noise ratio are fundamental in acoustics, and various
methods have been proposed in the past. If they are classified, (1) those utilizing directional
selectivity (directional microphones), (2) those utilizing differences in statistical properties
between signal and noise (various filters), (3) adaptive signals It is considered that it can be
classified into four types of processing technology (multi-channel noise removal method) and (4)
using spherical wave effect (close-talking microphone). In the case of the method (1), in the case
of an acoustic signal which can not but extend over a wide ratio band, the directivity actually
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obtained can not be sharpened, and at most it can not but be limited to bi-directionality and its
deformation degree. In general, the ability to improve the signal to noise ratio is very poor.
Furthermore, sharpening the directivity is possible in principle by making the overall size of the
microphone relatively large compared to the wavelength, but it becomes impossible to realize a
certain directivity over the entire frequency band . Noise usually has major energy in the low
frequency range. If this is to be dealt with, the dimensions of the microphone will be on the order
of a few meters, and it can not be used for practical use. Even if this is forced, the directivity
becomes too sharp at high frequency and it is not good condition.
In any case, it is difficult to achieve the purpose with the method (1), and in reality this method is
regarded as inappropriate. The method of (2) focuses on the so-called difference in statistical
properties between signal and noise, in particular the difference in power spectrum. The
principle limits of this method are given by the well-known Wiener filter, and it has been proved
in information theory that it is impossible to improve the signal to noise ratio beyond that. For
many acoustic signals we face, where the signal and noise spectra overlap, the limits of the
Wiener filter are quite small, providing only a very slight improvement of only a few dB at the
most. Therefore, it is not suitable at all for practical purposes where an improvement of about
several tens of dB is required. In the method of (3), the output of the microphone arranged in the
vicinity of the noise source to pick only the noise is passed through an adaptive filter having a
variable weighting factor, and the filter output is mixed with the signal and noise Decrease from
main microphone output. In order to minimize the difference output, in other words, to minimize
the noise component, the variable weighting coefficients of the adaptive filter are controlled
according to the algorithm to achieve the purpose. As an apparatus developed from this method,
there is one that uses a large number of microphones to sharply control the directivity. In any
case, the signal-to-noise ratio is improved to nearly 20 dB, and in the best case nearly 30 dB, so it
is far superior to the other methods. However, the problem is that the microphones have to be
placed near the noise source, and when adapting the directivity using a large number of
microphones, it is necessary to know in advance the distance and direction of the noise source.
These conditions are not always fulfilled, but rather will be overwhelmingly more unacceptable.
The disadvantages of this unrealistic limitation and the fact that the signal-to-noise ratio
improvement is still insufficient for some purposes. The method of (4) is in fact most often used
and has the advantage of being easy. However, the signal to noise ratio actually obtained is larger
than the method of (1) but smaller than the method of (3) because the change of the curvature of
the spherical wave which can not be obtained so much is used, but practically Although further
improvement is required, it is difficult in principle. [Problems to be Solved by the Invention] In
the above-described conventional signal-to-noise ratio improvement technique, although the
microphone used weights the signal and the external noise using directivity and other properties,
it is essentially linear. It is adding and converting into an electric signal.
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Therefore, sufficient performance can not be obtained with the methods (1) and (4) described
above (If directivity operation can be applied to the microphone output as non-linear operation,
the directivity is sharpened in any way. However, such methods can not be used for the following
reasons), and the methods (2) and (3) face extremely difficult information theory problems of
separating and removing noise once mixed in . The condition of this linearity is important and
must be fulfilled by any sound receiving system. This is not only because it is a universal
requirement in engineering, but also because in the sound receiving system nonlinear distortion
must be absolutely avoided. An object of the present invention is to solve the above problems and
to provide a sound receiving device capable of properly sounding a desired acoustic signal with a
large signal-to-noise ratio even in a noisy environment while maintaining linearity. [Means for
Solving the Problems] The sound receiving device of the present invention comprises sound
receiving elements disposed at a plurality of points on a virtual boundary surface surrounding
the central microphone as the sub microphone group, and the signal processing means The
sound pressure detection means for detecting the sound pressure of the external noise for each
sound receiving element, the first coefficient means for multiplying the value of the sound
pressure by a coefficient, and the sound pressure value multiplied by the coefficient Averaging
means for averaging over the entire surface, delay means for delaying the signal output from the
averaging means by a time corresponding to the distance between the central microphone and
the sound receiving element, and the central microphone And noise subtracting means for
subtracting from the output of The signal processing means further includes time differentiating
means for obtaining a time derivative of the sound pressure from the output of the sound
pressure detecting means, and second coefficient means for multiplying the value of the time
derivative by a coefficient, and the averaging means has a coefficient It is desirable to include
means for adding the multiplied time derivative value to the sound pressure value multiplied by
the factor. The signal processing means further includes a sound pressure gradient detection
means for detecting the normal direction sound pressure gradient of the boundary surface, and a
third coefficient means for multiplying the value of the sound pressure gradient by a coefficient,
and the averaging means It is desirable to include means for further adding the sound pressure
gradient value multiplied by the coefficient. The first to third coefficient means may be arranged
in the manual signal path or the output signal path of the respectively associated means. Also,
the delay means may be disposed anywhere on the signal path from the sound receiving element
to the noise subtraction means. The interface is preferably spherical. In order to divide it, it is
desirable to use a regular polyhedron.
In this case, since an equal amount of delay is given to the respective sound receiving elements,
one delay unit can be disposed downstream of the averaging means as the delay means. If it is
not a regular polyhedron, the coefficient values of the first to third coefficient means and the
delay amount of the delay means are appropriately selected. [Operation] The sound receiving
apparatus of the present invention focuses on space selectivity as a new fifth idea to avoid
extraneous noise while maintaining linearity. Here, the spatial selectivity is defined as a property
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that is sensitive only to the sound emitted from the sound source in a certain spatial area but
insensitive to the sound incident from the sound source outside the area. That is, the virtual
boundary surface is considered in space, and the outputs of the plurality of sub-microphones
disposed thereon and the main microphones placed at the ? -point in the boundary surface are
respectively appropriately signal processed to The performance is totally insensitive to the noise
coming from the In other words, spatial selectivity is obtained. At the same time, the sound
receiving device of the present invention is very sensitive to a sound source very close to the sub
microphone on the boundary surface, and a super close talk characteristic can be obtained. Also,
even when the submicrophone on the boundary surface is lacking, the hyper-close talk
characteristic can be obtained in the same direction as the lack. FIG. 1 is a block diagram of a
sound receiving apparatus according to a first embodiment of the present invention. This sound
receiving apparatus comprises a central microphone 100, a sub microphone group 200 disposed
around the central microphone 100, and a real time filter for removing extraneous noise
components included in the output of the central microphone 100 by the output of the sub
microphone group. And a signal processing circuit 300. The sub-microphone group 200 has a
plurality of points on an imaginary boundary surface surrounding the central microphone 100,
each surface of each of a regular M-face (in this embodiment, M = 4.6.8.12 or 20). It includes a
two-set of sound receiving elements 1-1.2-1 to 1-M, 2-M arranged at the center point. The signal
processing circuit 300 detects the sound pressure of the external noise and the value of the
sound pressure for each of the two sets of sound receiving elements 1-1.2-1 to 1-M and 2-M. As
a first coefficient means for multiplying the coefficients by a coefficient, adders 3-1 to 3-M and
coefficient multipliers 5-1 to 5-! 4 and the adder 10 and the coefficient multiplier 11 as
averaging means for averaging the sound pressure values multiplied by this coefficient over the
entire boundary surface, and the central microphone 100 receives the signal output from the
averaging means. A delay unit 12 is provided as a delay unit for delaying by a time
corresponding to the distance between the sound elements 1-1.2-1 to 1-M and 2-M, and the
output of the delayed averaging unit is output from the output of the central microphone 100. A
subtractor 13 is provided as noise subtraction means for subtraction.
The signal processing circuit 300 further includes differentiators 7-1 to 7-M as time
differentiation means for obtaining time differentiation of sound pressure from the output of the
sound pressure detection means, and a second coefficient for multiplying the value of this time
differentiation by a coefficient As a means, coefficient multipliers 8-1 to 8-) A are provided. In
addition, subtractors 4-1 to 4-M are provided as sound pressure gradient detection means for
detecting sound pressure gradients in the normal direction of the boundary surface, and
coefficients are provided as third coefficient means for multiplying the value of the sound
pressure gradient by a coefficient. Multipliers 6-1 to 6-M are provided. Corresponding to this
configuration, the averaging means includes adders 9-1 to 9-M as means for adding the time
differential value multiplied by the coefficient and the sound pressure gradient value to the
sound pressure value multiplied by the coefficient. . FIG. 2 shows the arrangement of the center
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microphone 100 and the sub microphone group 200 in the case of M = 12. The central
microphone 100 is disposed at the center of the regular dodecahedron, the sound receiving
elements 1-1 to 1-12 are disposed outside the central portion of each plane of the regular
dodecahedron, and the sound receiving elements 2-1 to 2- Each 12 is arranged inside the central
part of the corresponding surface. In FIG. 2, thirteen pairs of sound receiving elements 1-1.2-1 to
1-12.2-12 are shown as C0 to Cs and Do to D, respectively. Here, spatial selectivity will be
described. Since this is an unprecedented concept of sound reception, it is briefly explained
retrospectively to the wave equation that this is possible. FIG. 3 is a diagram for explaining the
principle of space selectivity, showing vectors at continuous boundary surface r and observation
point p. When there is a distribution ? (F, t) of the intensity of the sound source in the space, the
sound pressure ? (+ :, t) at point p in FIG. 3 satisfies the wave equation ииии (1). Where 7 is the
coordinate vector of point p and t is time. Starting from equation (1) in the region ? closed by
the virtual boundary surface r, using the green function G (i ?, t 1 i ? 01 t 0), ? (7 t) = J ? ?
? dtoff f, dvo [: p (r., to) c] ..... (2) is obtained. Here, dSo is a face vector. As a specific Green's
function, G (F, t l i ? ?, t,) = (1 / R) ? (R / c? (t?to)) relating to a three-dimensional free sound
field is adopted. If Equation (2) is modified by using this, + f ? ? dn dgoc ▒ 6 <R / c (t-to)>
gradograd4?-mar R иии (3) is obtained. Where ? (...) is the delta function of Dirac, ro and to are
the position vector and time for the sound source used in the definition of the Green function
(which does not necessarily coincide with the real sound source shown by 10 in FIG. 3) .
??????????????????? Assuming that the integration of the second term on
the right side of equation (3) is performed and the outward normal vector of the boundary
surface is no, dS = nodS according to the normal definition of the face vector, equation (3) is ,
(Following main page margin) = ? (r, L)-... (4) can be rewritten. The left side of equation (4)
means the sound coming from only the sound source in the closed space, from the definition of
the ? function. Sound coming from an external sound source, that is, no extraneous noise is
included. The arrangement of the central microphone 100 and the sub microphone group 200 is
two-dimensionally shown in FIG. In this arrangement, the central microphone 100 will sound the
sound pressure ? (F, t) from the sound source 40 in the area while the boundary surface r '(i'). )
6? / ? n 0, ? and a? / 6 ? ? of the external noise 41 by the sub-microphone group 200
appropriately distributed. The system can be made space selective if the signal processing is
performed according to the equation (4). However, in practice, it is not possible to measure 6? /
on, ?, ?, a? / at0 continuously. Therefore, for example, the boundary surface is equally divided
into M, a representative point is selected at the center, a sound receiving element is arranged at
that point, and a ? ? / ano, ?, a? / 6t0 at two representative points are obtained (4) It will
approximate the equation. In this case, the equation in the number-of-fee distribution receiving
element corresponding to the equation (4) is described with the left side of the equation (4)
abbreviated by the symbol ? (rq, t), ? (re, t): o (o) , T)--------(5). Here, the subscript 1 means the
quantity related to the i-th secondary microphone. ???????????????????
??????? Mountain и not) / Rt?Ci ? = fHdSot (Rt и not) / Rt??C. Here, L is a vector
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directed from the first sub microphone to the center microphone. Further, the surface
represented by the first sub microphone when the spherical surface is equally divided into M
pieces is designated rt. (5) The suffix t of = t?R, / c at the right end of the equation means that
the amount in the symbol ? is temporally earlier by t by Rt / c. That is, the amount in the symbol
? has to be time-adjusted through the delay of Ri / c after being observed and measured, and
must be subtracted from the output ? (o, t) of the central microphone. Since it is a simple
configuration to make the amount of delay constant, it is convenient to make the interface
spherical and place the central microphone at the center of the sphere.
If the boundary surface is selected to be a spherical surface and according to the division method
by a regular polyhedron, it is more convenient because al, bl and C become constant
independently of one. FIG. 5 is a diagram showing an XS'1SZ coordinate axis and a related vector
when a spherical surface of radius a is adopted as a boundary surface. 0 is the center of the
sphere, and in this case, aQ, which also serves as the origin, is a point sound source located at an
arbitrary position on the X axis (either inside or outside the sphere). ?? Is the position vector of
point Q, r is the vector from point O to a point on the sphere, To is the position of one point on
the sphere, and vector R is the vector from that point to the center of the sphere R = -ro, I R1 =
There is a relationship of l rol = a. Five. Is the unit vector in the normal direction going out of that
point on the sphere. Assuming that the radius of the sphere is a, then R = 1R1 = a, and five.
Because the direction of the and the moth is opposite, (? ? five. ???????? Therefore,
the sound velocity is C. Therefore, the equation (5) is ???qzt): ? (o, L) ? (1 / M) ? [a (a?t /
ano +) + ?; + (a / cXaQ; / atJ ? to = t?a / (....... The arrangement method of the sub
microphones satisfying the principle described above and the configuration for processing the
output of the sub microphones will be described. Also, the present invention starts from the idea
of achieving the purpose by arranging the sub-microphones on a closed arbitrary-shaped
boundary surface, so it is not limited to the spherical surface alone, but the delay and the weight
ai are indispensable in terms of construction. , Bi and C1 become constant independently of the
number i in the case of the spherical arrangement of the sub microphones, and it is judged that
the spherical arrangement method will actually be widely used, as it is an advantage in
constructing the device. First, the case of using a spherical surface as the boundary surface will
be specifically described. Divide the spherical surface into M by equal curved surface. For this
purpose, it is easy to think of starting from a regular polyhedron (although in geometry it is
known that there is a method of equal division other than regular polyhedrons). In a regular
polyhedron, only the case of M = 4.6.8.12.20 exists. The above-mentioned FIG. 2 shows the case
of the regular dodecahedron. The target sphere is inscribed in a regular dodecahedron, and is
equally divided into 12 in the plane passing through the center 0 and each ridge. According to
the equation (7), it is necessary to know the space differential a?i / ?not in the normal direction
of the sound pressure spherical surface. For this purpose, two minute sound receiving elements
may be disposed at two points separated by a minute distance ?r0 in the radial direction at each
sound receiving position, and the difference between the outputs of the two may be divided by
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?r0.
This is naturally performed by, for example, a velocity microphone, and is a well-known method.
The magnitude of ?r 0 may be selected to be a fraction of the wavelength of the highest
frequency to be used. The aforementioned first embodiment apparatus performs signal
processing in accordance with equation (7). If this content is demonstrated with reference to the
principle described above, first, the spherical surface is equally divided into M pieces. The sound
receiving elements 1-1 to 1-M are arranged slightly outside at the center position of the equally
divided surfaces, and the sound receiving elements 2-1 to 2-M are arranged slightly inside. The
sound receiving elements 1-1, 2-1.1-2 and 2-2, and so forth to 1-M and 2-1.4 M pairs of sound
receiving elements are only ?r 0 in the normal direction at each position. It has been put away.
The signal processing circuit 300 includes M signal processing units for each pair of sound
receiving elements. Since this signal processing unit has an identical structure, the signal
processing unit concerning the first sound receiving element 1-1.2-1 pair will be described. If the
output of the sound receiving element 1-1.2-1 is added by the adder 3-1 and multiplied by 172
by the coefficient multiplier 5-1, the output of the sound receiving element 1-1.2-1 is averaged
The sound pressure component ? 1 (iml) of the equation (7) is obtained. If this is further timedifferentiated by the differentiator 7-1 and multiplied by a / c by the coefficient multiplier 8-1,
the component (a / C) (a ? 1 / ? ja) II-1 related to the time differentiation of equation (7)
becomes It is obtained at the output of the coefficient multiplier 8-1. On the other hand, if the
difference of the creation output of the sound receiving element 1-1.2-1 is determined by the
subtractor 4-1 and multiplied by a / ?ro by the coefficient multiplier 6-1, the spatial
differentiation of the equation (7) can be obtained. The relevant term a (??t / ('not) l iml is
obtained. If these three types of components are added algebraically by the adder-subtractor 9-1,
then [a (a.noteq.t / [noi) +. Noteq.t + (a / c) (a <I'1 / <I>] of the equation (7) for iml 'to)] is
obtained. When the M similar signals obtained by each signal processing unit are added by the
adder 10 and multiplied by the coefficient multiplier 11 by 17) and quadrupled, the output is (L /
M) ? [a (a?r / ano, ) + ?1 + (a / cXa?; / at0) When this is further delayed by a / C by the delay
unit 12 and subtracted from the sound pressure output ? (o, t) of the central microphone 100
by the subtractor 13, ? (r ? 1) = ? (o, t) ? (L) / M) ? [: a (a ?; / ano +) + ? 1 + (a / c x a ? +
/ a Lo)]] L 6 ? -a / C- (7) is obtained and the object is achieved. FIG. 6 shows the space selection
characteristic of the apparatus of this embodiment. The vertical axis represents the output of this
device normalized as the output of the central microphone 10G with point sound tiles separated
by a distance rq as "l", ie, relative gain.
The horizontal axis KO is equal to (2? / ?) rq (where ? is the wavelength) and is the source
distance normalized with respect to the wavelength of the incident sound. The parameter is a
sphere radius equal to (2? / ?) a and normalized with respect to wavelength. Clearly, it can be
seen that the relative gain decreases as the source moves beyond the radius of the sphere. The
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sharpness of the cut is sufficiently sharp at about 35 dB per octave in this example of M = 12.
This trend increases with the value of M (11.5 dB per octave distance at M = 4). 37.5 dB per
octave distance at M-20). The lower limit of the blocking characteristic is obtained by KQ ? ?,
but the value increases as the normalized radius K increases, ie as the frequency or radius
increases. For example, assuming that a = 2.8 cm and frequency 4 kHz, M = 12 is obtained, and it
is expected that the relative gain is attenuated by 40 dB if the distance from the sound receiving
device is 40 cm. There is such a remarkable spatial selectivity characteristic. Also, as is clear from
the above description, this space selection characteristic is remarkable as the frequency is lower
and is easy to realize. This is the reverse of the sound devices known so far, and is a convenient
feature for the blocking of general noise which is of high energy in the low frequency range. FIG.
7 shows the change in relative gain when the sound source passes near the sub-microphone
placed at the interface. Since the secondary microphones are discretely present at the boundary
surface, advantageous effects other than the space selectivity or blocking characteristic described
in principle on the assumption of the continuous boundary surface previously occur. That is the
close talk characteristic described here. As shown in FIG. 7, it can be seen that the sensitivity is
clearly increased in the vicinity of the boundary. Therefore, this sound receiving device can
operate and be used as a so-called super-close-talking microphone that attenuates extraneous
noise significantly. FIG. 8 shows the spatial characteristics when one of the sound receiving
element pairs arranged at the interface is removed (M = ll). When a large number of sound
receiving element pairs are arranged at the boundary surface, the spatial selectivity can be made
directional by increasing the sensitivity of some sound receiving element pairs or decreasing the
sensitivity. In FIG. 8, the upper curve shows the space characteristic in the direction of the
missing and rough sound receiving element pair, and the lower curve shows the space selection
characteristic in the other direction. The relative gain increases in the direction lacking the sound
receiving element pair, and the attenuation is large in the other direction. In this case, even in the
direction in which the attenuation is large, the spatial selectivity is weaker than in the case of M =
12 in which the sound receiving element pair is uniform.
However, in practice, at the same time as having sufficient blocking characteristics, it can be used
as a close-talking microphone in the direction lacking the sub microphone. FIG. 9 is a block
diagram of a sound receiving apparatus according to a second embodiment of the present
invention. This example device is used when the shape of the interface is not spherical. The
difference from the first embodiment is that the coefficient multiplier 14-1 is inserted in the
signal path from the coefficient multiplier 5-i to the adder / subtractor 9-1 and the coefficient
values of the coefficient multipliers 6-i and 8-i are The difference is that the delay unit 12-1 is
inserted between the adder / subtractor 9-i and the adder 10 instead of the delay unit 12. When
the shape of the interface is not spherical, the value of i changes the coefficient term al SbL s C1
and the distance Ri from the interface to the central microphone 100. Therefore, as shown in the
second embodiment, the values of the coefficients of the coefficient multipliers of the respective
channels and the values of the delays of the delay units are adjusted to that. Specifically, the
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coefficient multiplier 5-1 of the i-th channel multiplies all the coefficients 172, and this output is
multiplied by-(1 / 4?) bI in the coefficient multiplier 14-1. The coefficient multipliers 6-i and 8-i
multiply by (1 / 47r) at ? ?r, = (1 / 4?) Ci, respectively. The delay unit 12-1 introduces delay
amounts R and / c. Although only the specific circuit configuration has been described in the
above embodiments, the present invention can be similarly implemented with any circuit
configuration as long as equivalent signal processing is performed. [Effects of the Invention] As
described above, the sound receiving device of the present invention can easily realize space
selectivity and close talk characteristics, and can adjust the desired signal with a high signal-tonoise ratio even in a noisy environment. It is extremely effective when applied to a humanpowered microphone for voice recognition and a microphone capable of preventing howling in
an acoustic system with loud sound.
[0002]
Brief description of the drawings
[0003]
FIG. 1 is a block diagram of a sound receiving apparatus according to a first embodiment of the
present invention.
FIG. 2 is a view showing an arrangement example of a central microphone and a sub microphone
group. FIG. 3 illustrates the principle of space selectivity. FIG. 4 is a view showing a twodimensional arrangement of a central microphone and a sub microphone group. FIG. 5 is a
diagram showing x, y, z coordinate axes and related vectors when a spherical surface of radius a
is adopted as a boundary surface. FIG. 6 is a view showing the space selection characteristic of
the apparatus of the embodiment. FIG. 7 is a view showing a change in relative gain when the
sound source passes near the sound receiving element arranged at the boundary surface. FIG. 8
is a diagram showing the space characteristic when one of the sound receiving element pairs
arranged at the interface is removed. FIG. 9 is a block diagram of a sound receiving device
according to a second embodiment of the present invention. 1-1 to 1-M, 2-1 to 2-M: sound
receiving element, 3-1 to 3-M: adder, 4-1 to 4-M: subtracter, 5- 1 to 5-M, 6-1 to 6-M, 8-1 to 8-M:
coefficient multiplier, 7-1 to 7-M ... differentiator, 9-1 to 9-M и и и Adder / subtractor 10 Adder 11
Coefficient multiplier 12.12-1 to 12-M-Delay 13-Subtractor 14-1 to 14-M Coefficient multiplier,
100: central microphone, 200: secondary microphone group, 300: signal processing circuit.
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