JP3170107B2 - Directional microphone system - Google Patents

Description

DETAILED DESCRIPTION OF THE INVENTION

[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 intended sound, such as voice input in a noisy environment,
In recent years, directional microphone systems have been used in order to perform sound collection with a high signal-to-noise ratio (SNR) between a target sound and ambient noise.

Conventionally, as such a directional microphone system, for example, a directional microphone system disclosed in Japanese Patent Application Laid-Open No. 4-278797 has been known.
In this directional microphone system, three non-directional microphone elements are arranged at regular intervals on a straight line, and a predetermined time delay processing and arithmetic processing are performed on the output from each microphone element to obtain strong directivity. It has become.

[0004]

In the conventional directional microphone system described above, strong directivity can be obtained with a simple configuration including only a delay circuit and a subtraction circuit, but the directivity pattern is determined by the arrangement of the microphone elements. And the sampling frequency, the directivity pattern cannot be changed according to the direction of the noise source.

According to the present invention, it is possible to increase a signal-to-noise ratio (SNR) of a sound to be collected and to easily change a directivity pattern according to a direction of a noise source, etc., with a simple configuration. It is intended to provide a possible directional microphone system.

[0006]

In order to achieve the above object, the invention according to claim 1 is provided in a predetermined linear direction.
And N omnidirectional sound receiving elements arranged at equal intervals, the N
A / D conversion means for A / D converting signals obtained by the plurality of sound receiving elements, and N-1 having the predetermined linear direction as a reference angle.
A / D conversion to eliminate sensitivity of sound wave from different angles
Means for each of the N signals A / D converted by the means.
Delay means for performing a fixed time delay process, and delay means
A predetermined sum for each of the N signals delayed by a predetermined time
Enter the summation means to be processed and values indicating N-1 angles.
Input means to input and N-1 angles from the input means
When a value is input, it indicates the input N-1 angles
A delay system for controlling a delay amount of the delay means based on the value.
And control means . Thus, under a simple configuration, noise from (N-1) directions is suppressed, and a signal-to-noise ratio (SNR) of a target sound to be collected is obtained.
Can be increased. Further, by changing the 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, and the like.
Even if the direction of the noise changes, this can be easily dealt with.

[0007]

According to a second aspect of the present invention, in the first aspect, an equalizer for correcting an amplitude characteristic is further provided. With this equalizer, it is possible to correct the attenuation of the amplitude characteristic in a low frequency band.

According to a third aspect of the present invention, a plurality of sets of the directional microphone system according to the second aspect are used for different continuous frequency bands to synthesize respective output signals. It is characterized by. Thus, 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 configuration diagram of an embodiment of a directional microphone system according to the present invention. Referring to FIG. 1, a directional microphone system includes a sound receiving unit including N sound receiving elements (specifically, non-directional microphone elements) M _{0 to} M _N−1 arranged at equal intervals on a straight line. An A / D converter 20 that converts the _N signals x _{0 to} x _{N−1 received} by the respective sound receiving elements M _{0 to} M _N−1 of the sound receiving unit 10 into digital signals;
A delay unit 50 for performing a predetermined delay process on each of the N signals digitally converted by the A / D conversion unit 20;
A summation unit 60 for performing a predetermined summation process on each of the N signals delayed by a predetermined time by 0.

The directional microphone system shown in FIG. 1 further includes an angle input unit 30 for inputting (N-1) values indicating angles, and an input from the angle input unit 30 (N-). 1) a delay control unit 40 for controlling the delay processing of the delay unit 50 based on the values indicating the angles.

In this directional microphone system, N
It is intended to eliminate the sensitivity of the sound coming from the (N-1) directions using the number of sound receiving elements (to make it “0”). when the the angle between the incident direction of the sound from the straight line direction and the sound source sound elements M ₀ to M _N-1 are arranged theta, to "0" the sensitivity of the angle theta (N-1 ) the number of the angle theta ₁ to theta _N-1 values theta ₁ to theta _N-1 corresponding to, and so as to input a value representing the (N-1) number of angles.

Further, the delay control unit 40 receives the (N-1) angles θ _n (n = 1 to 1) input from the angle input unit 30.
N−1) based on a value Θ _n (n = 1 to N−1)
2 ^N-1 delay amounts (including the number of delay amounts when the delay amount is “0”) are calculated. Further, the delay unit 5
0 has 2 ^N-1 pieces of delay circuits 2 ^N-1 pieces of delay amount calculated in the delay control section 40 are respectively set (not shown in FIG. 1), A / D converter unit the digital converted each of the N signals at 20, 2 ^N-1 pieces of delay with 2 ^N-1 pieces of delay amount set in the circuit, thereby performing a predetermined delay processing .

Here, Θ _n (n = 1 to N−1) is a value (integer) represented by the following equation with respect to the angle θ _n (n = 1 to N−1), as described later. Numerical value) can be used.

[0015]

Θ _n = (τ / T) cos θ _n

In the above equation, τ is an acoustic delay time,
T is, for example, a sampling cycle of A / D conversion in the A / D conversion unit 20, and θ _n is treated as a discrete amount.

Next, the principle of the directional microphone system of the present invention will be described. FIG. 2 is a diagram showing a relationship between the N sound receiving elements (omnidirectional microphone elements) M _{0 to} M _{N -1} of the sound receiving unit 10 and the sound source. In FIG. 2, N sound receiving elements M _{0 to} M _N−1 are arranged at regular intervals d (= τc,
τ is the acoustic delay time, and c is the speed of sound). Now, it is assumed that the sound from the sound source is a plane wave having an angular frequency ω. An input signal x _n (n = 0 to N−1) obtained as a result of sound reception by each sound receiving element M _n (n = 1 to N−1) is expressed by the following equation with respect to time t and angle θ. It is.

[0018]

X _n (t, θ) = x ₀ (t, θ) · exp (−jnωτ · cos θ)

Here, the output signal of the system is represented by y (t,
θ), and the transfer function of the system Η (ω, θ) = y (t,
θ) / x ₀ (t, θ), Η (ω, θ) in the following equation is considered.

[0020]

(Equation 3)

[0021] Number 3 transfer function Η (ω, θ) is "0" when theta = theta _n, therefore, as a system, build realizes the transfer function of the number 3 Eta the (omega, theta) Then, the output signal y becomes “0” at the angle θ _n (n = 1 to N−1).
Next, cancel the sound from the N-1 angle _{θ n (n = 1~N-1} ) direction, it is possible to eliminate the sensitivity to angle _{θ n (n = 1~N-1} ) sound from a direction .

Therefore, in equation ₍ 2), cos θ _n is calculated as
_When approximated by _n , Equation 3 can be transformed as follows.

[0023]

(Equation 4)

From equation (4), Η (ω, θ) is expressed as exp (−jωτco
sθ) can be expanded as a polynomial and can be expressed as the following equation.

[0025]

(Equation 5)

Here, f _n (exp (jωT)) is expressed by the following equation.

[0027]

(Equation 6)

Here, S is N-1 Θ (Θ ₁ , Θ ₂ ,
…, Θ _N-1 ) to any n Θ (Θ _m1 , Θ _m2 , ..., Θ _mn )
Is selected and summed. Also, ΣΣ ... Σ is N
It corresponds to all combinations when n 場合 are selected from Θ. Therefore, the output signal y (t, θ) of the system
Can be expressed as: Therefore, the output signal y (t, θ) of the system can be expressed as:

[0029]

(Equation 7)

The physical meaning of Equation 7 is that each input signal x _n (t,
θ), a delay operation of f _n (exp (jωT)) is performed, and a sum of the respective delay operation results f _n (exp (jωT)) x _n (t, θ) is obtained to obtain an output signal y ( t, θ).
At this time, an effective delay operation f _n (exp (ωT)) · x _n (t, θ)
, S must not be positive in equation (6). That is, when S is positive, the delay amount cannot be obtained. Therefore, since it is assumed the S non-positive value, using the maximum value theta _max that can be taken of S, each of f _n (e
Change uniformly in the direction of delaying xp (jωT)) only further T.theta _max, as the output signal of the system, y a (t, θ) T
A signal y (t−t) expressed by the following equation delayed by Θ _max
TΘ _max, θ) is to be output.

[0031]

(Equation 8)

[0032] Thus, theta = theta _n transfer function of "0" so that the number 3 at the time of (Η (ω, θ)) is represented by the number 7, the output signal represented by the number 7 y (t, θ)
For (N-1) pieces of angle theta _n sound from the direction "0"
Next, also, y (t, theta) was allowed simply T.theta _max delay y
(t−TΘ _max , θ) also becomes “0” for (N−1) sounds from the angles θ _n . Therefore, if the system is capable of performing the arithmetic processing of Equation 8, the sensitivity of sound from (N-1) angles can be eliminated.

The directional microphone system shown in FIG. 1 has a configuration capable of easily performing the arithmetic processing of Expression 8. That is, when the user inputs _N−1 integer values Θ _{1 to} Θ _N−1 corresponding to the angles at which the sensitivity is desired to be lost from the angle input unit 30, the delay control unit 40 sets these Θ ₁
To take the sum of any combination of ^{_{Θ N-1 (2 N-}} 1 single sum), a 2 ^N-1 pieces of delay, 0, theta _1, theta _{2,
Θ 3, Θ 1 + Θ 2} , Θ 2 + Θ 3, Θ 1 + Θ 3, ..., Θ 1 + Θ 2 +
Θ ₃ +... + Θ _N−1 , and comparing these, the largest one of them is Θ _max
Calculate as Then, the delay control unit 40 calculates the difference between each of the 2 ^N−1 delay amounts and Θ _max and sets them as regular delay amounts to the 2 ^N−1 delay circuits of the delay unit 50 ( (Not shown in FIG. 1). In this state, A / D
N input signals x _n (t, θ) (n = 0 to 0) from the conversion unit 20
N-1) is added to the delay unit 50, the delay unit 50 uses 2 ^{N -1} delay circuits and a predetermined number of adder circuits for each input signal x _n (t, θ), and _n '(exp (jωT)) x
A delay operation process of _n (t, θ) is performed. When the delay unit 50 obtains N delay operation results f _n 'exp (jωT)) · x _n (t, θ) for the N input signals x _n (t, θ),
These are sent to the summation unit 60. The summation unit 60, seeking the sum of these (including a difference not only Japanese), which can be output as _{y (t-TΘ max, θ} ).

As described above, in this embodiment, the delay control unit 4
0, a predetermined number of addition circuits and a predetermined number of subtraction circuits are provided, respectively, the delay unit 50 is provided with 2 ^{N -1} delay circuits and a predetermined number of addition circuits, and the summation unit 60 is provided with an addition / subtraction circuit ( (In consideration of the sign operation of (-1) ⁿ , a subtraction circuit is also included.) The arithmetic processing of Expression 8 can be easily realized with a simple configuration.

FIG. 3 shows four (N = 4) sound receiving elements M _0.
Or M ₃ is provided in three directions of angles theta _1, theta _2, specific examples of the directional microphone system when eliminating a sensitivity with respect to sounds from the theta ₃ is shown. FIG. 4 shows a configuration example of the delay unit 50 and the summation unit 60 in the directional microphone system of FIG. In the configuration example of FIG. 4, the delay unit 50 includes eight delay circuits 101 to 108 and four addition circuits 109 to 112, and the summation unit 60 includes a code corresponding to the order of the microphone elements. Adder / subtractor circuits 113 to 115 which perform addition / subtraction according to
It is constituted by.

In the system of FIG. 3, the angles θ ₁ ,
When the user digitally inputs the integer values θ ₁ , Θ ₂ , Θ ₃ corresponding to these angles from the angle input unit 30 in order to eliminate the sensitivity to the sounds of θ ₂ and θ ₃ , these integer values Θ ₁ ,
Θ ₂ and Θ ₃ are added to the delay control unit 40 and the delay control unit 40
Then, the sum for any combination of Θ ₁ , Θ ₂ , Θ ₃ ;
_{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 is determined as the delay amount (in this case, 2 ^N-1 = 8 delay amounts).

The delay control unit 40 calculates the maximum of the eight delay amounts as Θ _max , and
Θ _max is subtracted from each of the delay amounts to obtain −Θ _max , Θ ₁
−Θ _max , Θ ₂ −Θ _max , Θ ₃ −Θ _max , Θ ₁ + Θ ₂ −Θ _max ,
Θ ₁ + Θ ₃ −Θ _max , Θ ₂ + Θ ₃ −Θ _max , Θ ₁ + Θ ₂ + Θ ₃ −
Θ _max is determined as a delay amount (eight delay amounts), and these are obtained as 2 ^{N -1} (= 8) delay circuits 101, 102, 103, 104, 105, 106, and 106 in the delay unit 50 shown in FIG. 1
07 and 108 respectively.

The input signal x ₀ ~x ₃ from four sound receiving devices M ₀ to M ₃ of the sound receiving element portion 10 is A / D converted by the A / D converter 20, when applied to the delay unit 50 , Delay unit 50
Then, the four signals x ₀ to x transmitted from the A / D conversion unit 20 are _output .
relative x _3, based on the eight delay amount set to the eight delay circuit _{101~108, f n '(Z)} · x n (n = 0~
The delay calculation processing of 3) is performed. Here, jωT is set to Z for simplicity. In Equation 8, when N is 4,
f _n ′ (Z) · x _n (n = 0 to 3) can be represented by the following equation.

[0039]

(Equation 9)

Actually, for example, the distance d (= 4) between four sound receiving elements M _{0 to} M ₃ which are arranged at equal intervals on a straight line.
τc) is 85 mm (sound speed c is 340 m / sec, τ = 1/4 m)
Second), the sampling frequency in the A / D converter 20 is 13.33 kHz (T = 3/40 msec), and three angles θ ₁ ,
The cosines cosθ ₁ , cosθ ₂ and cosθ ₃ of θ ₂ and θ ₃ are
Assuming that “−0.3”, “0.6”, and “−0.9”, Θ ₁ , Θ ₂ , and Θ ₃ are “−1”,
These values are “2” and “−3”, and these values are
From the user. In the delay control unit 40,
_1, theta _2, theta ₃ input values "-1", "2", "- 3" based on the eight delay _{0, Θ 1, Θ 2,} Θ 3, Θ 1 + Θ 2, Θ 1
+ Θ ₃ , Θ ₂ + Θ ₃ , Θ ₁ + Θ ₂ + Θ ₃ are “0”,
"-1", "2", "-3", "1", "-4", "-
1 ”and“ −2 ”, and the largest of these, ie, Θ _max , is determined as“ 2 ”.
The delay control unit 40 calculates 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 ₃ −
Θ _max , Θ ₂ + Θ ₃ −Θ _max , Θ ₁ + Θ ₂ + Θ ₃ −Θ _max are respectively “−2”, “−3”, “0”, “−5”, “−
1 ”,“ −6 ”,“ −3 ”, and“ −4 ”, and these are calculated as delay circuits 101, 102, 103,
104, 105, 106, 107 and 108 are set.
In this case, f _n ′ (Z) · x _n in Expression 9 is represented by the following equation.

[0041]

Equation 10] _{f 0 '(Z) x 0} = + Z -2 · x 0 f 1' (Z) x 1 = - (Z -3 + Z 0 + Z -5) · x 1 f 2 '(Z) x 2 ^{= + (Z -1 + Z -6} + Z -3) · x 2 f 3 '(Z) x 3 = -Z -4 · x 3

In the delay section 50, eight delay circuits 101 to
The input signals x _{0 to} x ₃ are delayed based on the eight delay amounts set at 108. That is, the delay circuit 101 delays the input signal x ₀ by Z ⁻² . Further, the delay circuit 102, 103 and 104, to the input signal x ₁ Z ^-3, straining facilities delays Z ^0, Z ^-5. Also, delay circuit 1
05, 106 and 107, the input signal x ₂
^-1 , Z ^-3 and Z ^-6 are applied. The delay circuit 108
In, straining facilities delays Z ^-4 input signal x _3. Delay circuit 1
The output results from 02, 103 and 104 are added by the adders 109 and 110 of the delay unit 50, and f ₁ ′ (Z) × ₁
= (Z ⁻³ + Z ⁰ + Z ⁻⁵ ) × ₁ and the output results from the delay circuits 105, 106, 107 are added to the adder 1
11 and 112, and f ₂ ′ (Z) × ₂ = (Z ⁻¹ +
Obtained as _{^{Z -3 + Z -6) x 2}} . On the other hand, the delay circuit 10
The output result from 1 is obtained as f ₀ ′ (Z) x ₀ = Z ⁻² x ₀ , and the output result from the delay circuit 108 is f ₃ ′ (Z)
is obtained as x ₃ = Z ^-4 x _3. As is clear from Equation 10, f ₁ ′ (Z) x ₁ and f ₃ ′ (Z) x ₃ are negative (−).
Requires a sign.

The summation unit 60 includes the delay circuit 1 of the delay unit 50.
Output f ₀ from 01 'and ^{_{(Z) x 0 = Z -2}} x 0, output from the adding circuit _{109,110 f 1' (Z) x} 1 = (Z -3 +
Z ⁰ + Z ^-5 ) x ₁ , output result f ₂ ′ (Z) x ₂ = (Z ^-1 + Z ^-3 + Z ^-6 ) x ₂ from adders 111 and 112, and output result from delay circuit 108 Four delay results of f ₃ ′ (Z) x ₃ = Z ⁻⁴ x ₃ are added, and the addition and subtraction circuits 113 to 115 add
The sum of these values is calculated. At this time, the output results, that is, (Z ⁻³ + Z ⁰ + Z ⁻⁵ ) x ₁ and Z ⁻⁴ x ₃ are given negative signs (that is, subtracted). Thereby, the summation unit 60
Can output an output signal y (t-2).

FIGS. 5 and 6 show a directional characteristic (directivity pattern) and an amplitude characteristic of the output signal y (t-2), respectively. FIG. 5 shows the directivity characteristics when ωτ = π / 2 and the sampling frequency is 3.33 kHz, and FIG. 6 shows the amplitude characteristics when the angle θ is “0”. It is. As can be seen from FIG. 5, the output y (t−
2) is cos θ ₁ = −0.3, cos θ ₂ = 0.6, cos θ ₃ =
It is insensitive to sound waves (plane waves) from the directions of _three angles θ ₁ , θ ₂ , and θ ₃ of −0.9.
In other words, the directional pattern can be changed according to the direction of the noise source by appropriately changing 指向₁ , Θ ₂ , and Θ ₃ . Further, it can be seen that the output y (t-2) has a narrow directivity with high sensitivity in a certain direction, that is, in the present example, in the forward direction (θ = 0). From this, the directional microphone system of the present embodiment can easily change the directional pattern according to the direction of the noise source, etc., with a simple configuration, and achieve the purpose of collecting sound. The signal-to-noise ratio (SNR) of the sound can be increased.

As can be seen from the amplitude characteristics at the angle θ = 0 shown in FIG. 6, the sensitivity is attenuated at low frequencies. As shown in FIG. 7, the system shown in FIG. 1 (FIG. 3) includes:
An equalizer 70 for correcting the amplitude characteristic can be further added. In this case, a low-pass filter having a cutoff frequency of 330 Hz and a slope of −18 dB / oct can be used as the equalizer 70.

Further, a plurality of sets of the microphone system of FIG. 7 are used for different continuous frequency bands, respectively.
It is also possible to configure such that a strong directional characteristic is provided over a wide frequency band as a whole. FIG. 8 shows a specific example of a directional microphone system that uses two sets of the microphone systems of FIG. 7 and covers two frequency bands of a low frequency band and a high frequency band. That is, the directional microphone system of FIG.
A sound receiving unit 210 provided with _{0 to} M ₅ and an A / D conversion unit 220
, The angle input unit 230, and the first and second delay control units 24
0a, 240b and first and second delay units 250a, 250
0b, first and second summation sections 260a and 260b, first and second equalizers 270a and 270b, and synthesis section 2
80 are provided.

Here, for the low frequency band, the first element interval d
The four sound receiving elements M ₀ , of the sound receiving unit 210 arranged at ₁
M ₁ , M ₂ , M ₃ , a first delay control unit 240a,
, A first summation section 260a, and a first equalizer 270a, and a fourth section of the sound receiving section 210 arranged at a _second element interval d2 for high frequencies.
Two sound receiving elements M ₀ , M ₄ , M ₅ , M ₂ , a second delay control unit 240b, a second delay unit 250b, and a second summation unit 2
60b and a second equalizer 270b.

In such a configuration, when each output from the six sound receiving elements M ₀ , M ₁ , M ₂ , M ₃ , M ₄ , M ₅ of the sound receiving section 210 is applied to the A / D conversion section 220. , A / D converter 220
Then, the outputs from the four sound receiving elements M ₀ , M ₁ , M ₂ , and M ₃ are A / D-converted at a low-frequency sampling frequency (1 / T ₁ ) and provided to the first delay unit 250a. Also, the sound receiving unit 210
A / D-convert the outputs from the _four sound receiving elements M ₀ , M ₄ , M ₅ , and M _{2 at} a sampling frequency (1 / T ₂ ) for a high frequency band, and provide the result to a second delay section 250b. First delay unit 250a
Then, the output from the A / D conversion unit 220 is subjected to a predetermined delay processing with a predetermined delay amount set by the first delay control unit 240a, the sensitivity is corrected by the first equalizer 270a, and the synthesis unit 280 is processed. Give to. Also, the second delay unit 250
In b, a predetermined delay process is applied to 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 sensitivity is corrected by the second equalizer 270b. This is provided 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 the output of the entire system.

Now, the first element interval d ₁ for the low band is set to 85.
mm, the sampling frequency (1 / T ₁ ) for the low band is 1
And 3.33KHz, first equalizer 270a cutoff frequency 330 Hz, b properties of -18 dB / oct - constituted by-pass filter, also the second element spacing d ₂ 28.3 mm for high-frequency, high The sampling frequency (1 / T ₂ ) for the frequency band is set to 40 kHz, and the second equalizer 27 is used.
0b is cutoff frequency 1kHz, -18dB / oct
, A strong directional characteristic can be provided in a wide frequency band from 330 Hz to 3 kHz in the entire system.

In the above-described embodiment, the delay unit 50 and the summation unit 60 are configured as shown in FIG. 4. However, instead of the configuration example shown in FIG. 4, these are configured as shown in FIG. It is also possible. That is, the delay unit 50 includes only eight delay circuits 101 to 108, and the summation unit 60 includes seven add / sub circuits that add / subtract the delay result from each of the delay circuits 101 to 108 in accordance with the sign according to the order of the microphone elements. 121 to 128 can also be used. Further, the delay unit 50 and the summation unit 60 can be configured as one instead of being separate.

Further, in the above-described embodiment, θ
was Θ _n taking _n of the cosine of the (cos), but the cosine (co
instead of s), it can also be a θ _n taking the sine (sin) Θ of _n.

[0052]

As described above, according to the first aspect of the present invention, N elements arranged at equal intervals in a predetermined linear direction.
And number of non-directional sound receiving device, and the A / D converting means for a signal obtained by said N pieces of sound receiving element for converting A / D, said predetermined linear
Sensation of sound wave from N-1 angles with line direction as reference angle
A / D conversion by A / D conversion means to eliminate the degree
A predetermined time delay process is applied to each of the N signals obtained.
And to delay means, N delayed a predetermined time by the delay means
Summing means for performing a predetermined summation process on each of the signals, input means for inputting values indicating N-1 angles, and input means.
When a value indicating N-1 angles is input from the force means,
Based on the input values indicating the N-1 angles, the delay
Delay control means for controlling the delay amount of the extension means.
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 with a simple configuration. Also, N-1 pieces
Can be input, and N-1 angles can be input.
When a value indicating the degree is input, the delay
Since the delay amount of the extension means is controlled, noise
It is possible to change the directivity pattern according to the direction of the sound source, etc.
Yes, if the direction of the noise changes,
Can be easily dealt with.

[0053]

According to the second aspect of the present invention, since the equalizer for correcting the amplitude characteristic is further provided, it is possible to correct the attenuation of the amplitude characteristic in a low frequency band.

Further, according to the third aspect of the present invention, wherein
Since a plurality of sets of the directional microphone system according to item 2 are used for different continuous frequency bands and their output signals are combined, strong directivity can be provided in a wide frequency band. .

[Brief description of the drawings]

FIG. 1 is a configuration diagram of an embodiment of a directional microphone system according to the present invention.

FIG. 2 is a diagram illustrating a relationship between N sound receiving elements of a sound receiving unit and a sound source.

FIG. 3 is a diagram illustrating a specific example of a directional microphone system.

FIG. 4 is a diagram illustrating a configuration example of a delay unit and a summation unit of the directional microphone system of FIG. 3;

FIG. 5 is a diagram showing the directional characteristics of output signals of the directional microphone systems of FIGS. 3 and 4;

FIG. 6 is a diagram showing an amplitude characteristic of an output signal of the directional microphone system of FIGS. 3 and 4;

FIG. 7 is a configuration diagram of a directional microphone system in which an equalizer is further provided in the configuration of FIG. 1;

8 is a diagram showing a configuration example of a directional microphone system using two sets of the directional microphone systems of FIG. 7;

9 is a diagram illustrating another configuration example of the delay unit and the summation unit of the directional microphone system of FIG. 3;

[Explanation of symbols]

 Reference Signs List 10 sound receiving unit 20 A / D conversion unit 30 angle input unit 40 delay control unit 50 delay unit 60 summation unit 101 to 108 delay circuit

Claims (3)

(57) [Claims] 1. An N array having equal intervals arranged in a predetermined linear direction.
And number of non-directional sound receiving device, and the A / D converting means for a signal obtained by said N pieces of sound receiving element for converting A / D, said predetermined linear
Sensation of sound wave from N-1 angles with line direction as reference angle
A / D conversion by A / D conversion means to eliminate the degree
A predetermined time delay process is applied to each of the N signals obtained.
And to delay means, N delayed a predetermined time by the delay means
Summing means for performing a predetermined summation process on each of the signals, input means for inputting values indicating N-1 angles, and input means.
When a value indicating N-1 angles is input from the force means,
Based on the input values indicating the N-1 angles, the delay
A directional microphone system comprising: a delay control unit that controls a delay amount of the extension unit . 2. The directional microphone system according to claim 1, further comprising an equalizer for correcting an amplitude characteristic. 3. A directional microphone system, wherein a plurality of sets of the directional microphone system according to claim 2 are used for different continuous frequency bands to synthesize respective output signals. .