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JPH0723494

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DESCRIPTION JPH0723494
[0001]
BACKGROUND OF THE INVENTION 1. Field of the Invention The present invention relates to a
directional microphone system used for inputting voice and music.
[0002]
2. Description of the Related Art When there is noise in the surroundings other than the target
sound, such as voice input under a noise environment, sound collection with a high signal-tonoise ratio (SNR) between the target sound and the ambient noise In recent years, directional
microphone systems have been used to
[0003]
Conventionally, as such a directional microphone system, for example, a directional microphone
system disclosed in JP-A-4-278797 is known.
In this directional microphone system, three nondirectional microphone elements are arranged at
equal intervals on a straight line, and predetermined time delay processing and arithmetic
processing are performed on outputs from the respective microphone elements to obtain strong
directivity. It is supposed to be.
03-05-2019
1
[0004]
In the above-described conventional directional microphone system, strong directivity can be
obtained with a simple configuration of only the delay circuit and the subtraction circuit, but the
directivity pattern is not limited to the arrangement of the microphone elements. And the
sampling frequency, and there is a problem that the directivity pattern can not be changed
according to the direction of the noise source and the like.
[0005]
According to the present invention, it is possible to increase the signal-to-noise ratio (SNR) of a
sound to be collected and to easily change the directivity pattern according to the direction of the
noise source, etc., with a simple configuration. It is an object of the present invention to provide a
sex microphone system.
[0006]
SUMMARY OF THE INVENTION In order to achieve the above object, the invention according to
claim 1 comprises N nondirectional sound receiving elements arranged at equal intervals along a
straight line, and the N A / D converting means for A / D converting a signal obtained by the
sound receiving element of claim 1, 2N-1 delaying means for delaying the A / D converted signal
for a predetermined time, and a signal obtained from the delaying means And summing means
for determining the sum of
This makes it possible to suppress noise from (N-1) directions and increase the signal-to-noise
ratio (SNR) of the target sound to be picked up under a simple configuration.
[0007]
In the inventions according to claims 2 to 4, in the invention according to claim 1, an input
means for inputting values indicating N-1 angles, and a value indicating N-1 angles from the
input means And the delay control means for controlling the delay amount of the delay means on
the basis of these values, and a value indicating N-1 angles from the input means. It is possible to
change the directivity pattern according to the direction of the noise source, etc. by changing the
value of .beta., And this can be easily dealt with even when the direction of the noise changes.
[0008]
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2
The invention according to claim 5 is characterized in that, in the invention according to claim 1,
an equalizer for correcting the amplitude characteristic is further provided.
This equalizer can correct the attenuation of the amplitude characteristic in the low frequency
band.
[0009]
The invention as claimed in claim 6 is characterized in that plural sets of the directional
microphone system according to claim 5 are used for different continuous frequency bands to
synthesize respective output signals. There is.
Thereby, strong directivity can be provided in a wide frequency band.
[0010]
Embodiments of the present invention will be described below with reference to the drawings.
FIG. 1 is a block diagram of an embodiment of a directional microphone system according to the
present invention. Referring to FIG. 1, the directional microphone system includes a sound
receiving unit 10 having N sound receiving elements (specifically, nondirectional microphone
elements) M0 to MN-1 arranged at equal intervals along a straight line. A / D conversion unit 20
for converting N signals x0 to xN-1 received by the respective sound receiving elements M0 to
MN-1 of the sound receiving unit 10 into digital signals; A delay unit 50 which applies a
predetermined delay process to each of the N converted signals, and a summing unit 60 which
applies a predetermined summation process to each of N signals delayed by a predetermined
time by the delay unit 50. And have.
[0011]
Further, the directional microphone system of FIG. 1 further includes (N-1) angle input units 30
for inputting values indicating (N-1) angles and (N-1) input values from the angle input units 30.
And a delay control unit 40 that performs control relating to the delay processing of the delay
unit 50 based on the value indicating the angle of.
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3
[0012]
In this directional microphone system, it is intended that N sound receiving elements be used to
cancel ("0") the sensitivity of sound coming from (N-1) directions, and therefore, When the angle
between the linear direction in which the sound receiving elements M0 to MN-1 are arranged
and the incident direction of the sound from the sound source is θ, the sensitivity of the angle
input unit 30 is “0” in the angle θ. Values Θ1 to ΘN-1 corresponding to (N-1) pieces of
angles θ1 to θN-1 are input as values indicating (N-1) pieces of angles.
[0013]
Further, the delay control unit 40 is based on values Θ n (n = 1 to N−1) indicating (N−1) pieces
of angles θ n (n = 1 to N−1) input from the angle input unit 30. Thus, 2N-1 delay amounts
(including the number when the delay amount is “0”) are calculated.
The delay unit 50 has 2N-1 delay circuits (not shown in FIG. 1) to which 2N-1 delay amounts
calculated by the delay control unit 40 are respectively set. The predetermined delay processing
is performed on each of the N signals digitally converted by the D conversion unit 20 using 2N-1
delay amounts set in the 2N-1 delay circuits. There is.
[0014]
Here, as 後 述 n (n = 1 to N−1), as described later, for the angle θ n (n = 1 to N−1), use a value
(integer value) represented by the following equation Can.
[0015]
[Equation 1] Θ n = [(τ / T) cos θ n]
[0016]
In the above equation, τ is an acoustic delay time, T is, for example, a sampling period of A / D
conversion in the A / D conversion unit 20, and [•] means a function in which • is an integer.
[0017]
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4
Next, the principle of the directional microphone system of the present invention will be
described.
FIG. 2 is a view showing the relationship between the N sound receiving elements (nondirectional
microphone elements) M0 to MN-1 of the sound receiving unit 10 and the sound source.
In FIG. 2, the N sound receiving elements M0 to MN-1 are arranged on a straight line at equal
intervals d (= τc, τ is acoustic delay time, c is sound velocity).
Now, it is assumed that the sound from the sound source is a plane wave of angular frequency ω.
An input signal xn (n = 0 to N-1) obtained as a result of being received by each sound receiving
element Mn (n = 1 to N-1) is expressed by the following equation with respect to time t and angle
θ.
[0018]
xn (t, θ) = x0 (t, θ) · exp (−jnωτ · cosθ)
[0019]
Here, the output signal of the system is y (t, θ), and the transfer function of the system Η (ω,
θ) = y (t, θ) / x0 (t, θ). )think of.
[0020]
The transfer function Η (ω, θ) of Eq. 3 becomes "0" when θ = θ n, and hence a system which
realizes the transfer function Η (ω, θ) of Eq. For example, the output signal y is “0” when the
angle θ n (n = 1 to N−1), and cancels the sound from the direction of the N−1 angles θ n (n =
1 to N−1), The sensitivity to the sound from the direction of the angle θ n (n = 1 to N−1) can
be eliminated.
[0021]
Then, in the equation 2, when cos θ n is approximated by Θ n represented by the equation 1,
the equation 3 can be transformed as the following equation.
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5
[0022]
From Eq. 4, Η (ω, θ) can be expanded as a polynomial of exp (-jωτ cos θ) and can be
expressed as the following equation.
[0023]
Here, fn (exp (jωT)) is expressed by the following equation.
[0024]
Thus, the output signal y (t, θ) of the system can be expressed as:
[0025]
The physical meaning of Eq. 7 is calculated by delaying each input signal xn (t, θ) by fn (exp
(jωT)) and calculating each delay operation result fn (exp (jωT)) xn The sum of (t, θ) is obtained
to obtain an output signal y (t, θ).
At this time, S must not be positive in Eq. 6 in order to make a valid delay operation fn (exp (ωT))
× n (t, θ).
That is, when S is positive, the delay amount can not be obtained.
Therefore, in order to make S non-positive, each fn (exp (jωT)) is further uniformly delayed by T
遅 延 max using the maximum value Θmax that S can take, and the output signal of the system is
obtained. Then, a signal y (t−TΘmax, θ) represented by the following equation obtained by
delaying y (t, θ) by TΘmax is output.
[0026]
Thus, the transfer function (Η (ω, θ)) of Eq. 3 which becomes “0” when θ = θ n is
approximately represented by Eq. 7 and represented by Eq. Output signal y (t, θ) becomes “0”
03-05-2019
6
with respect to the sound from (N−1) angles θn, and y (t, θ) is simply delayed by TΘmax y
(t−TΘmax) , Θ) also become “0” for sounds from (N−1) angles θ n directions.
Therefore, in the case of a system capable of performing the arithmetic processing of Formula 8,
it is possible to eliminate the sensitivity of the sound from (N-1) angular directions.
[0027]
The directional microphone system shown in FIG. 1 is configured to be able to easily perform the
arithmetic processing of Equation 8.
That is, when the user inputs N-1 integer values Θ1 to 対 応 N-1 corresponding to the angle at
which the user wants to lose sensitivity from the angle input unit 30, the delay control unit 40
arbitrarily combines these Θ1 to ΘN-1 Take the sum (2N-1 sums) of, and calculate 2N-1 delay
amounts as 0, Θ 1, Θ 2, Θ 3, Θ 1 + Θ 2, Θ 2 + Θ 3, Θ 1 + Θ 3, ..., Θ 1 + Θ 2 + Θ 3 + ... + Θ N
-1 The largest of these is determined as Θmax.
Then, the delay control unit 40 takes the difference between each of the 2N-1 delay amounts and
Θmax, and uses them as the normal delay amount, in the 2N-1 delay circuits of the delay unit 50
(see FIG. (Not shown) is set.
In this state, when N input signals xn (t, θ) (n = 0 to N-1) are added to the delay unit 50 from the
A / D conversion unit 20, the delay unit 50 generates each input signal xn (t , Θ) are subjected to
fn ′ (exp (jωT)) × x (t, θ) using 2N−1 delay circuits and a predetermined number of addition
circuits. When N delay operation results f n 'exp (jω T)) x n (t, θ) for N input signals x n (t, θ)
are obtained in the delay unit 50, these are sent to the summing unit 60. . The summation unit
60 can obtain the summation (including not only the summation but also the difference) and
output it as y (t−TΘmax, θ).
[0028]
Thus, in this embodiment, the delay control unit 40 is provided with 2N-1 addition circuits and
subtraction circuits, and the delay unit 50 is provided with 2N-1 delay circuits and a
predetermined number of addition circuits. In addition, the arithmetic processing of Equation 8
can be easily realized with a simple configuration in which only the addition / subtraction circuit
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7
(also including the subtraction circuit in consideration of the sign operation of (−1) n) is
provided in the summing unit 60. can do.
[0029]
FIG. 3 shows a specific example of a directional microphone system in which four (N = 4) sound
receiving elements M0 to M3 are provided and the sensitivity to sound from angles θ1, θ2, and
θ3 in three directions is eliminated. It is done.
Further, FIG. 4 shows a configuration example of the delay unit 50 and the summing unit 60 in
the directional microphone system of FIG. 3. In the configuration example of FIG. 4, the delay
unit 50 is configured by eight delay circuits 101 to 108 and four adder circuits 109 to 112, and
the summing unit 60 is a code according to the order of the microphone elements. It comprises
three addition / subtraction circuits 113 to 115 which perform addition / subtraction according
to.
[0030]
In the system of FIG. 3, when the user digitally inputs integer values Θ1, Θ2 and Θ3
corresponding to these angles from the angle input unit 30, in order to eliminate sensitivity to
sounds of angles θ1, θ2 and θ3 in three directions, These integer values Θ 1, Θ 2 and Θ 3
are added to the delay control unit 40, and the delay control unit 40 sums 0, 組 み 合 わ せ 1, Θ
2, Θ 3, Θ 1 + Θ 2, Θ 2 + Θ 3, and Θ 1 + Θ 3 in arbitrary combinations of Θ 1, Θ 2 and Θ 3.
Θ1 + Θ2 + Θ3 is obtained as a delay amount (in this case, 2N-1 = 8 delay amounts).
[0031]
Further, the delay control unit 40 indexes the largest one of the eight delay amounts as Θmax,
subtracts Θmax from each of the eight delay amounts, and obtains −Θmax, Θ1-Θmax, Θ2Θmax. , Θ3-Θmax, Θ1 + Θ2-Θmax, Θ1 + Θ3-Θmax, Θ2 + Θ3-Θmax, Θ1 + Θ2 + Θ3-Θmax
are determined as normal delay amounts (eight normal delay amounts), 2N-1 of the delay unit 50
shown in FIG. The delay circuits 101, 102, 103, 104, 105, 106, 107, and 108 are respectively
set.
[0032]
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8
When the input signals x0 to x3 from the four sound receiving elements M0 to M3 of the sound
receiving element unit 10 are A / D converted by the A / D conversion unit 20 and added to the
delay unit 50, the delay unit 50 Fn '(Z) .xn (n = n) based on the eight delay amounts set in eight
delay circuits 101 to 108 for the four signals x0 to x3 sent from / D conversion unit 20. Perform
delay calculation from 0 to 3).
Here, jωT is set to Z for simplicity.
In Formula 8, when N is 4, fn '(Z) xn (n = 0 to 3) can be expressed as the following formula.
[0033]
Actually, for example, the distance d (= τc) between the four sound receiving elements M0 to
M3 arranged at equal intervals on a straight line is 85 mm (sound speed c is 340 m / s, τ = 1/4
m) Sec), and the sampling frequency in the A / D conversion unit 20 is 13.33 KHZ (T = 3/40 ms),
and cosines θ1, θ2 and θ3 of three angles θ1 and θ2 desired to be desensitized Assuming
that cos θ2 and cos θ3 are respectively “−1.3” “0.6” and “−0.9”, according to
Equation 1, Θ1, Θ2 and −13 are respectively “−1”, “2” and “− These values are input
by the user from the angle input unit 30. In the delay control unit 40, eight delay amounts 0, ”1,
Θ2,, 3, Θ1 + Θ2, Θ1 + Θ3, Θ2 + Θ3, Θ1 + Θ2 + Θ3 based on the input values" -1 "," 2 "and" 3 "of Θ1, Θ2 and Θ3. Is calculated as “0”, “−1”, “2”, “−3”, “1”, “−4”, “−1”,
“−2” and the largest of these, That is, Θmax is indexed as "2". Next, the delay control unit 40
sets the delay amount 0, Θ 1-Θ max, Θ 2-Θ max, Θ 3-Θ max, Θ 1 + Θ 2-Θ max, Θ 1 + Θ 3Θ max, Θ 2 + Θ 3-Θ max, Θ 1 + Θ 2 + Θ 3-Θ max as "-2", "- Calculated as 3 "," 0 "," -5 "," -1 ","
-6 "," -3 "," -4 ", and these are calculated as the delay circuits 101, 102, 103, 104 of the delay unit
50. , 105, 106, 107, and 108 are set. In this case, fn ′ (Z) · xn of equation 9 is expressed as the
following equation.
[0034]
f0'(Z)x0=+Z-2・x0f1'(Z)x1=−(Z-3+Z0+Z-5)・x1f2'(Z)x2=+(Z-1+Z6+Z-3)・x2f3'(Z)x3=−Z-4・x3
[0035]
03-05-2019
9
The delay unit 50 delays the input signals x0 to x3 based on the eight delay amounts set in the
eight delay circuits 101 to 108.
That is, in the delay circuit 101, the input signal x0 is delayed by Z-2. The delay circuits 102,
103, and 104 apply delays of Z-3, Z0, and Z-5 to the input signal x1, respectively. The delay
circuits 105, 106, and 107 respectively delay the input signal x2 by Z-1, Z-3, and Z-6. The delay
circuit 108 delays the input signal x3 by Z-4. The output results from the delay circuits 102, 103,
and 104 are added by the adder circuits 109 and 110 of the delay unit 50, and f1 '(Z) x1 = (Z-3 +
Z0 + Z-5) x1 is obtained. , 106, and 107 are added by the adder circuits 111 and 112 to obtain f2
'(Z) x2 = (Z-1 + Z-3 + Z-6) x2. On the other hand, the output result from the delay circuit 101 is
obtained as f0 '(Z) x0 = Z-2x0, and the output result from the delay circuit 108 is obtained as f3'
(Z) x3 = Z-4x3. As is clear from Equation 10, a negative (-) code is required for f1 '(Z) x1 and f3'
(Z) x3.
[0036]
In the summing unit 60, the output result f0 '(Z) x0 = Z-2x0 from the delay circuit 101 of the
delay unit 50 and the output result f1' (Z) x1 = (Z-3 + Z0 + Z- from the adder circuits 109 and
110). 5) x1 and the output result f2 '(Z) x2 = (Z-1 + Z-3 + Z-6) x2 from the adder circuits 111 and
112, and the output result f3' (Z) x3 = Z-4x3 from the delay circuit 108 And add / subtract
circuits 113 to 115 to calculate the sum of them, but in this case, the output result, that is, (Z-3 +
Z0 + Z-5) x1 and Z-4x3 are given negative signs. (Ie subtract). Thereby, the output signal y (t-2)
can be output from the summing unit 60.
[0037]
5 and 6 respectively show the directivity characteristic (directivity pattern) and the amplitude
characteristic of the output signal y (t-2). FIG. 5 shows directivity characteristics when ωτ = π /
2 and the sampling frequency is 3.33 KHZ, and FIG. 6 shows amplitude characteristics when the
angle θ is “0”. It is. As can be seen from FIG. 5, the output y (t-2) is a sound wave from the
direction of three angles θ1, θ2, θ3 with cos θ1 = −0.3, cos θ2 = 0.6, cos θ3 = −0.9. There
is no sensitivity to (plane wave). In other words, the directivity pattern can be changed according
to the direction of the noise source by appropriately changing the ridges 1, 2, and 3. Also, it can
be seen that the output y (t-2) has narrow directivity with high sensitivity in a certain direction,
that is, in the forward direction (θ = 0) in this example. From this, in the directional microphone
system of this embodiment, it is possible to easily change the directivity pattern according to the
03-05-2019
10
direction of the noise source and the like with a simple configuration, and to collect the sound.
The signal to noise ratio (SNR) of the sound can be increased.
[0038]
As can be seen from the amplitude characteristic in the case of the angle θ = 0 shown in FIG. 6,
the sensitivity is attenuated at low frequencies. In order to prevent (correct) the sensitivity
attenuation at low frequencies, an equalizer 70 for correcting the amplitude characteristic can be
further added to the system of FIG. 1 (FIG. 3) as shown in FIG. In this case, as the equalizer 70, a
low pass filter having a cutoff frequency of 330 Hz and a slope of -18 dB / oct can be used.
[0039]
Alternatively, a plurality of sets of microphone systems shown in FIG. 7 may be used for different
continuous frequency bands to provide strong directivity over a wide frequency band as a whole.
FIG. 8 shows an example of a directional microphone system which uses two sets of the
microphone system of FIG. 7 and covers two frequency bands of low frequency and high
frequency band. That is, the directional microphone system of FIG. 8 includes a sound receiving
unit 210 including six sound receiving elements M0 to M5, an A / D conversion unit 220, an
angle input unit 230, and first and second delay control. The units 240a and 240b, the first and
second delay units 250a and 250b, the first and second summing units 260a and 260b, the first
and second equalizers 270a and 270b, and the combining unit 280 are provided. It is done.
[0040]
Here, for the low band, the four sound receiving elements M0, M1, M2, M3 of the sound
receiving unit 210 arranged at the first element interval d1, the first delay control unit 240a, and
the first delay Section 250a, a first summing section 260a, and a first equalizer 270a, and the
four sound receiving elements M0 of the sound receiving section 210 arranged at a second
element interval d2 for high frequency band use. , M4, M5, and M2, a second delay control unit
240b, a second delay unit 250b, a second summing unit 260b, and a second equalizer 270b.
[0041]
In such a configuration, when each output from the six sound receiving elements M0, M1, M2,
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M3, M4, and M5 of the sound receiving unit 210 is applied to the A / D conversion unit 220, the
A / D conversion unit 220 The outputs from the four sound receiving elements M0, M1, M2 and
M3 are A / D converted at the sampling frequency (1 / T1) for the low frequency band and given
to the first delay unit 250a. The outputs from the four sound receiving elements M0, M4, M5,
and M2 are A / D converted at the sampling frequency (1 / T2) for the high frequency and
supplied to the second delay unit 250b.
In the first delay unit 250a, the output from the A / D converter 220 is subjected to
predetermined delay processing with a predetermined delay amount set by the first delay control
unit 240a, and the sensitivity correction is performed by the first equalizer 270a. To the
synthesis unit 280. In addition, the second delay unit 250b performs predetermined delay
processing on the output sent from the A / D conversion unit 220 with a predetermined delay
amount set by the second delay control unit 240b, and the second equalizer The sensitivity is
corrected at 270 b and applied to the combining unit 280. The combining unit 280 combines the
output from the first equalizer 270a and the output from the second equalizer 270b, and outputs
this as an output of the entire system.
[0042]
Now, the first element distance d1 for low frequency is 85 mm, the sampling frequency (1 / T1)
for low frequency is 13.33 KHz, and the first equalizer 270a has a cutoff frequency of 330 Hz
and -18 dB / oct. The second element interval d2 for the high frequency band is 28.3 mm, the
sampling frequency for the high frequency band (1 / T2) is 40 KHz, and the second equalizer
270b has a cutoff frequency of 1 KHz. In the case of a low-pass filter with -18 dB / oct
characteristics, the entire system can have strong directivity for a wide frequency band from 330
Hz to 3 KHz.
[0043]
Although the delay unit 50 and the summing unit 60 are configured as shown in FIG. 4 in the
above-described embodiment, they may be configured as shown in FIG. 9 instead of the
configuration example of FIG. It is.
That is, seven adding / subtracting circuits that configure the delay unit 50 with only eight delay
circuits 101 to 108 and add / subtract the delay results from the delay circuits 101 to 108
according to the code according to the order of the microphone elements It can also consist of
121-128. Furthermore, the delay unit 50 and the summing unit 60 may not be separate but may
03-05-2019
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be combined into one.
[0044]
In the above-described embodiment, the cosine (θ) of θn is taken in Equation 1 to obtain Θn,
but instead of the cosine (cos), the sine of θn may be taken to be Θn.
[0045]
As described above, according to the first aspect of the present invention, N nondirectional sound
receiving elements arranged at equal intervals on a straight line, and the N sound receiving
elements are used. A / D conversion means for A / D converting the obtained signal, 2N-1
delaying means for delaying the A / D converted signal for a predetermined time, and summation
for finding the sum of the signals obtained from the delaying means And means for suppressing
noise from (N-1) directions and enhancing a signal-to-noise ratio (SNR) of a target sound to be
picked up under a simple configuration.
[0046]
Further, according to the invention of claims 2 to 4, it is possible to input values indicating N-1
angles, and when values indicating N-1 angles are input, based on these. Since the delay amount
of the delay means is controlled, the directivity pattern can be changed according to the direction
of the noise source, etc., and even when the direction of the noise changes, Can be easily dealt
with.
[0047]
Further, according to the fifth aspect of the present invention, since the equalizer for correcting
the amplitude characteristic is further provided, the attenuation of the amplitude characteristic in
the low frequency band can be corrected.
[0048]
Further, according to the invention of claim 6, since plural sets of the directional microphone
system according to claim 5 are used for different continuous frequency bands, respective output
signals are synthesized. Strong directivity can be provided in a wide frequency band.
[0049]
Brief description of the drawings
03-05-2019
13
[0050]
1 is a block diagram of an embodiment of a directional microphone system according to the
present invention.
[0051]
2 is a diagram showing the relationship between the N sound receiving elements of the sound
receiving unit and the sound source.
[0052]
3 is a diagram showing a specific example of a directional microphone system.
[0053]
4 is a diagram showing an example of the configuration of the delay unit and the summing unit
of the directional microphone system of FIG.
[0054]
5 is a diagram showing the directional characteristics of the output signal of the directional
microphone system of FIG. 3, FIG.
[0055]
6 is a diagram showing the amplitude characteristics of the output signal of the directional
microphone system of FIG. 3, FIG.
[0056]
7 is a configuration diagram of a directional microphone system further provided with an
equalizer in the configuration of FIG.
[0057]
8 is a diagram showing a configuration example of a directional microphone system using two
sets of the directional microphone system of FIG.
[0058]
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9 is a diagram showing another configuration example of the delay unit and the summing unit of
the directional microphone system of FIG.
[0059]
Explanation of sign
[0060]
DESCRIPTION OF SYMBOLS 10 Sound receiving part 20 A / D conversion part 30 Angle input
part 40 Delay control part 50 Delay part 60 Sum total part 101-108 Delay circuit
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