JPH06261390A - Microphone - Google Patents

Abstract

PURPOSE:To provide the microphone obtaining a good sound collecting quality even in sound collecting sound fields with much acoustic echoes near the diffusion sound field. CONSTITUTION:The device is provided with a directional 1st microphone 11, an omni-directinal 2nd microphone 21, comparison means 24 comparing the output levels between the 1st and 2nd microphones 11 and 21, and high--ass band filter 13 provided for the output of the 1st microphone 11. A device output signal is taken out from the fiter 13. Based on the output of the comparison means 24, the roll-off frequency of the high-pass band filter 13 is controlled to change according to the level ratio of the outut of the 1st and 2nd microphones 11 and 21.

Description

Detailed Description of the Invention

[0001]

BACKGROUND OF THE INVENTION 1. Field of the Invention The present invention relates to a microphone device which can be widely used, for example, a microphone mounted on a camera-integrated VTR, a microphone for speech, a microphone for recording a conference, and the like.

[0002]

2. Description of the Related Art In order to eliminate background sound as much as possible and collect only desired sound, a microphone having directivity such as unidirectionality is often used.

There is also an attempt to apply an adaptive noise reduction device to eliminate unnecessary noise and obtain only desired voice. About the adaptive noise reduction device,
This will be described below.

In FIG. 10, 1 is a main input terminal, 2 is a reference input terminal, and a signal input through the main input terminal 1 is supplied to a combining circuit 4 via a delay circuit 3. The signal input through the reference input terminal 2 is
It is supplied to the combining circuit 4 via the adaptive filter circuit 5 and subtracted from the signal from the delay circuit 3. The output of the synthesis circuit 4 is fed back to the adaptive filter circuit 5 and is also led to the output terminal 6.

In this noise reduction device, a desired signal s and a noise signal n0 uncorrelated with the desired signal s are input to the main input terminal 1.
The sum of and is input. On the other hand, the noise signal n1 is input to the reference input terminal 2. The noise signal n1 of this reference input is designed to be uncorrelated with the desired signal s, but correlated with the noise signal n0.

The adaptive filter circuit 5 filters the reference input noise signal n1 and outputs a signal y that approximates the noise signal n0. Output signal y of this adaptive filter circuit 5
As an alternative, it is also possible to obtain a signal having a phase opposite to that of the noise signal n0 and an equal amplitude. The delay circuit 3 is for adjusting the time of the signals to be subtracted in consideration of the processing time in the adaptive filter circuit 5.

The adaptive algorithm in the adaptive filter circuit 5 works so as to minimize the subtraction output (residual output) e which is the output of the synthesis circuit 4. That is, in the adaptive filter circuit 5, the update amount of the adaptive filter coefficient is obtained from the residual output e and the reference input by the calculation means, and the adaptive filter coefficient is updated.

Assuming that s, n0, n1 and y are statistically stationary and the average value is 0, the residual output e is e = s + n0-y. The expected value of the square of this is that s is n0,
Also, because it is y ^{uncorrelated, E [e 2] = E} [s 2] + E [(n0 -y) 2] + 2E [s (n0 -y)] = E [s 2] + E [(n0 - y) ² ]. Assuming that the adaptive filter circuit 5 converges,
The adaptive filter circuit 5 updates the adaptive filter coefficient so that E [e ² ] is minimized. At this time, since E [s ² ] is not affected, Emin [e ² ] = E [s ² ] + Emin [(n 0 -y) ² ].

[0009] That is, E E by [e ^2] is minimized [(n0 -y) ^2] is minimized, the output y of the adaptive filter circuit 5 will estimate of the noise signal n0.
The expected value of the output from the synthesis circuit 4 is the desired signal s
Will only be. That is, adjusting the adaptive filter circuit 5 to minimize the total output power is equivalent to the subtraction output e becoming the least-squares estimated value of the desired speech signal s.

The adaptive filter circuit 5 may be realized by either an analog signal processing circuit or a digital signal processing circuit.

In this adaptive noise reduction apparatus, as shown in FIG. 10, the output audio signal of the main input microphone 7 is supplied to the main input terminal 1 and the output audio signal of the reference input microphone 8 is supplied to the reference input terminal 2. Supply.

Further, in this case, when the desired voice arrival direction is AR, the main input microphone 7 is
An omnidirectional microphone as shown in 1A is used. Alternatively, a unidirectional microphone is arranged with its main directional axis facing the desired voice arrival direction.

The reference input microphone 8 is shown in FIG.
As shown in FIG. 1B, the unidirectional microphone has sensitivity in the voice input direction to be excluded, and the desired voice arrival direction A
It is used by arranging it so that R is in the directivity null direction.
When the arrival direction of the voice to be excluded is 90 degrees with respect to the desired voice arrival direction, the bidirectional microphone is arranged and used so that the desired voice arrival direction AR is the directional null direction. You may

According to the microphone device to which such an adaptive noise reduction device is applied, when the adaptive noise reduction operation is performed, a signal extremely close to the desired voice is obtained as the device output, which is substantially equivalent to the super directional microphone. The above good sound collection quality can be realized as a small device instead of a large device such as a super-directional microphone.

[0015]

By the way, even if the microphone having the above-mentioned directivity is designed so that the sound pressure-frequency characteristic becomes flat in a free sound field such as outdoors,
When this microphone is used to collect sound in a reverberant sound field (reverberant sound field or diffuse sound field) such as in a room, the low range of surrounding voice and noise is exaggerated, and it is very difficult to hear. It is known to become difficult.

In order to deal with the frequency characteristic of such a directional microphone, when the environment in which the microphone is used is predetermined, the microphone is adjusted so that the sensitivity in the low frequency range becomes low according to the environment in which it is used. Is being designed. However, since this is not versatile, a configuration is also adopted in which the microphone device is provided with a switch for manually switching the frequency characteristic so that the user can appropriately select the frequency characteristic according to the usage environment.

However, in the method of manually switching the frequency characteristic of the microphone with the switch, it is difficult in operation to finely switch the frequency characteristic depending on the sound pickup field, and normally one of two switching modes is used. You only get the freedom to choose.

In many cases, the user may forget to switch the switch, and the prepared frequency characteristic switching mode may not be useful.

Further, in the case of a microphone device using a VTR with a built-in camera, it is difficult to switch the switch when the sound is picked up while moving from the outdoor to the indoor (or vice versa). In addition, many noises are mixed by switching the switch during sound collection, and sudden changes in sound quality may be unnatural.

When a microphone device using the adaptive noise reduction device is used, when sound is picked up in a free sound field or a sound field with less reverberation, a sound wave incident on the microphone unit is directly incident on it. Since it can be considered as only sound waves, the output level of the microphone 7 is greater than or equal to the output level of the microphone 8.

However, when sound is picked up by a microphone device using an adaptive noise reduction device in a sound field having a lot of reverberation close to a diffuse sound field, the microphones 7 and 8 are indirectly connected to sound waves other than directly incident sound waves. Incident sound waves increase and sometimes indirect sounds are more common. In such a situation, the output level of the microphone 7 becomes lower than the output level of the microphone 8, and this becomes remarkable especially in the low range due to the frequency characteristics of the directional microphone unit in the reverberant sound field. .

As described above, the sound field to be used is diffuse, the sound is increased, and the level of background noise (the total noise of the system existing regardless of the presence or absence of the transmission signal) is equal to or higher than the level of the desired input voice. Becomes large, the property of the adaptive noise reduction device is that the rejection rate of unnecessary noise is limited, and it becomes difficult to obtain good sound collection quality.

A first object of the present invention is to solve the above problems by automatically changing the frequency characteristic of the output of a microphone having directivity according to the sound pickup field.

A second object of the present invention is to reduce the deterioration of the sound collection quality in a sound field with a lot of reverberation, which is close to the diffuse sound field, in the microphone device using the adaptive noise reduction device.

[0025]

[Means for Solving the Problems] Even when a microphone is designed so that its frequency characteristic becomes flat in a free sound field, in a reverberant sound field or a diffuse sound field, an omnidirectional microphone and a directional microphone are used. In the case of, the frequency characteristic changes. That is, in the case of an omnidirectional microphone, the frequency characteristics hardly change from the free sound field even in the reverberant sound field or the diffuse sound field, but in the case of the directional microphone, the sound field in the reverberant sound field or the diffuse sound field. The sensitivity becomes higher in the bass range.

The present invention utilizes the fact that the directional microphone and the omnidirectional microphone have different frequency characteristics in the sound field used, as described above.
And to achieve the second object.

That is, the microphone device according to claim 1 of the present application corresponds to the reference numerals of the embodiment shown in FIG. 1 described later, and the directional first microphone 11 and the omnidirectional second microphone are associated with each other. 21, a comparison means 24 for comparing the output levels of the first and second microphones 11 and 21, and a high-pass filter 13 provided for the output of the first microphone 11; The device output signal is taken out from 13 and the roll-off frequency of the high-pass filter 13 changes based on the output of the comparison means 24 according to the level ratio of the outputs of the first and second microphones 11 and 21. It is characterized by doing so.

In order to achieve the above second object,
The microphone device according to claim 2 of this application corresponds to the reference numeral of the embodiment of FIG. 8 described later, and the first microphone 61 for picking up a desired voice and the arrival direction of the desired voice. The second microphone 71 having a low sensitivity and directivity, the adaptive filter means 74 to which the audio signal from the second microphone is supplied, and the output signal of the adaptive filter means 74 from the audio signal of the first microphone 61. A subtracting means 65 for subtracting, a means for adjusting the adaptive filter means 74 so that the output power of the subtracting means 65 is minimized, a directional third microphone 71,
An omnidirectional fourth microphone 61, a level ratio detecting means 81 for detecting the level ratio of the output audio signals of the third and fourth microphones, and a level ratio detected by the level ratio detecting means 81 and a predetermined value. Means for limiting the adaptive operation of the adaptive filter means 74 based on the deviation from the value.

[0029]

In the reverberation sound field or the diffuse sound field, the fact that the sound field sensitivity of the directional microphone becomes high in the low sound range will be described below by taking the case of the condenser microphone as an example.

FIG. 12 shows an equivalent circuit of the condenser microphone. In FIG. 12, S0 is the stiffness of the diaphragm, R0 is the viscous resistance between the diaphragm and the back pole, M0 is the mass of the diaphragm, S1 is the stiffness of the back air chamber, R1 is the braking resistance of the rear entrance hole, and P0. Is the sound pressure at the front of the diaphragm, and P1 is the sound pressure at the rear of the unit.

If the distance between the two acoustic terminals of the front surface of the diaphragm and the rear entrance hole of this microphone is d and the area of the diaphragm is S, the relationship between the sound pressure P0 and the sound pressure P1 with respect to the sinusoidal incident sound wave. Is expressed by the following equation (1).

[0032]

[Equation 1] Here, ω = 2πf is the angular frequency, C is the speed of sound, and θ is the angle of incidence. At this time, the displacement Xs of the diaphragm with respect to the incident sound pressure from the front is expressed by the following equation (2).

[0033]

[Equation 2] On the other hand, in the diffuse sound field, the sound pressure levels of the sound waves incident on both input terminals of the front surface of the diaphragm and the rear entrance hole are equal,
They are uncorrelated with each other. The sound pressure incident on the front surface of the diaphragm is N
If the sound pressure at the rear of the unit is 0 and the sound pressure at the rear of the unit is N1, then the displacement Xd of the diaphragm at this time is expressed by the following equation (3).

[0034]

[Equation 3] When the displacement of the diaphragm in the diffuse sound field is normalized by the displacement of the diaphragm with respect to the sinusoidal incident sound wave, it is expressed by the following equation (4).

[0035]

[Equation 4] In this equation (4), if α = R1 · C / S1 · d (5), the equation (4) can be described as the following equation (6).

[0036]

[Equation 5] Here, α is a quantity representing directivity, and when α = 0, it is bidirectional, when α = 1, it is unidirectional, and when α = ∞, it is omnidirectional.

From the above equation (6), it is possible to know the frequency characteristic in the diffuse sound field when the sound pressure frequency characteristic of the microphone unit in the free sound field is flat.

For example, when the unit is omnidirectional,
Since α = ∞, it can be seen that the frequency characteristic does not depend on the frequency, that is, the frequency characteristic is flat.

Further, when the unit is unidirectional, since α = 1, the high range is flat, but the sensitivity is such that the sensitivity rises in inverse proportion to the frequency in the low range. . When the unit is bidirectional,
Since α = 0, it can be seen from Equation (6) that the frequency characteristic is similar to that of unidirectionality.

Although the case of the condenser microphone has been described above, the same characteristics are exhibited in the case of other types of microphones.

The present invention utilizes the characteristics of the microphone described above. That is, the directional first
And the omnidirectional second microphone have the same frequency characteristics in the free sound field, the sound pickup field of the microphone device according to the present invention is a free sound field or a sound field with less reverberation. In this case, since the sound to be collected may be considered to be only the sound wave directly incident on the microphone, the level ratio of both microphones input to the comparison means has a predetermined value corresponding only to the difference in directivity. Therefore, when the comparing means detects that the deviation between the level ratio and the predetermined value is small, the sound collecting sound field is recognized as a free sound field or a sound field having a small reverberation.

On the other hand, when the picked-up sound field is a sound field with a lot of reverberation close to the diffuse sound field, as described above, the directional first field is used.
The microphone has high sound pressure sensitivity in the low frequency range, and the level ratio of the output of both microphones input to the comparison means is
It changes from the time of the free sound field. That is, the deviation between the level ratio and the predetermined value becomes large, and the comparison means recognizes that the sound pickup sound field is a sound field with a lot of reverberation and is close to a diffuse sound field.

In this way, the sound collecting sound field is recognized from the level ratio of the outputs of both microphones, the roll-off frequency of the high-pass filter is controlled to an appropriate value, and the output sound signal of good sound collecting quality is obtained. Is what you get.

[0044]

1 is a block diagram showing the configuration of an embodiment of a microphone device according to the present invention.

Reference numeral 11 is a first microphone for obtaining a picked-up voice output signal, which is a directional microphone having a flat sound pressure-frequency characteristic in a free sound field, for example. As the microphone 11, a unidirectional microphone, a bidirectional microphone, or a superdirective microphone can be used. The output audio signal of the first microphone 11 is led to the output terminal 14 via the amplifier 12 and the high pass filter 13.

Reference numeral 21 is a second microphone for recognizing the sound collecting sound field together with the first microphone. As the second microphone 21, a first microphone in a free sound field is used.
An omnidirectional microphone having the same sound pressure-frequency characteristic as that of the microphone 11 in this example, which has a flat sound pressure-frequency characteristic is used.

The output audio signal of the second microphone 21 is supplied to the level detection circuit 23 via the amplifier 22 and the level is detected. Then, the level detection signal is supplied to the level comparison circuit 24. Further, the output signal of the first microphone 11 from the amplifier 12 is supplied to the level detection circuit 25, the level is detected, and the level detection signal is supplied to the level comparison circuit 24.

The level comparison circuit 24 determines the level ratio between the output level of the first microphone 11 and the output level of the second microphone 21. The level ratio detected by the comparison circuit 24 is supplied to the control signal forming circuit 26. The control signal forming circuit 26 forms a control signal for determining the roll-off frequency of the high pass filter from the level ratio.

In the case of this example, the high-pass filter 13
Is composed of a capacitor 131 and a transistor 132 as a variable impedance element, and the base voltage of the transistor 132 is supplied from the control signal forming circuit 26 as a control signal.

In the above configuration, when the sound pickup sound field is a free sound field, the microphone 11 and the microphone 21 pick up only the sound that is directly incident on it.
The level ratio of the outputs of both microphones 11 and 21 detected by the comparison circuit 24 becomes a value close to the level ratio which is a predetermined value corresponding to the directivity of both microphones which is known in advance. At this time, a voltage that makes the transistor 132 non-conductive is supplied to the transistor 132 as a control signal. Therefore, the high-pass filter 13 has a roll-off frequency of 0 or a very low frequency, and the output audio signal of the microphone 11 is guided to the output terminal 14 with almost the same frequency characteristic.

Further, when the picked-up sound field is a sound field with a lot of reverberation close to the diffuse sound field, the output level of the microphone 11 becomes large as described above, and the level comparison circuit 24
At, it is detected. That is, the level ratio has a deviation of a predetermined value or more from the predetermined value, and the control signal forming circuit 26 forms a base voltage according to the deviation between the obtained level ratio and the predetermined value as a control signal. As a result, the roll-off frequency of the high pass filter 13 changes to the higher frequency side according to the magnitude of the deviation.

Therefore, when the picked-up sound field is a sound field with a lot of reverberation close to the diffuse sound field, the low-pass level of the microphone 11 becomes high, but the roll-off frequency of the high-pass filter 13 becomes higher. Due to the change of, the output audio signal obtained at the output terminal 14 can have a substantially flat frequency characteristic, and an output audio signal with good sound collection quality can be obtained.

FIG. 2 is a configuration block diagram showing another embodiment of the microphone device according to the present invention. In this example, low-pass filters 27 and 28 are inserted between the amplifier 12 and the level detection circuit 25 and between the amplifier 22 and the level detection circuit 23, respectively, and the microphone 1
The level ratios of only the low frequency components of the output signals 1 and 21 are compared and detected. Then, the high-pass filter 13 is controlled in the same manner as in the above example according to the level ratio. Other configurations are exactly the same as the example of FIG.

As described above, since the change in the frequency characteristic due to the difference in the picked-up sound field is mainly seen in the low frequency range, the configuration of the example of FIG. 2 improves the accuracy of the level ratio detection / comparison. , More appropriate operation can be obtained.

In the example of FIG. 2, band pass filters may be used instead of the low pass filters 27 and 28.

FIG. 3 is another example of the microphone device according to the present invention, which is different from the example of FIG. 1 in the configuration of the high-pass filter 13 and the control method of this filter, and shows the attenuation characteristics of the high-pass filter. The steepness is changed.

That is, the high-pass filter 13 of this example
Are respectively composed of a capacitor C and a transistor Tr as a variable impedance element, and have three high-pass filters 133 and 1 with a roll-off frequency of 100 Hz, for example.
34 and 135 are connected in cascade. Then, the output signal of the comparison circuit 24 is supplied to the control signal forming circuit 29, from which the filter 13
A control signal is supplied to the bases of the transistors Tr of 3,134,135.

In this example, when the picked-up sound field is a free sound field, the level ratio detected by the comparison circuit 24 is
The value is close to the default value, and the control signal forming circuit 29 is set to 3
All transistors T of one filter 133-135
A control signal is supplied to r so that it becomes non-conductive. Then, the output audio signal of the microphone 11 through the amplifier 12 is guided to the output terminal 14 with almost the same frequency characteristic.

When the picked-up sound field is not a free sound field and the reverberation increases and the output level of the second microphone 21 increases, the comparison circuit 24 detects that as the output level ratio of both microphones 11 and 21. To be done. Then, the control signal forming circuit 29 controls so that the steepness of the attenuation characteristic of the high-pass filter 13 increases as the output level of the first microphone 11 increases as compared with the output level of the second microphone 21. To do.

That is, in a sound field with little reverberation, the level ratio of both microphones 11 and 21 has a small deviation from the default value in the free sound field. At this time, the base voltage is supplied from the control signal forming circuit 29 to the transistor Tr of the filter 133 so that the transistor Tr has a predetermined impedance such that the roll-off frequency becomes 100 Hz, and the other filters 134 and 135. Transistor Tr
Is supplied with a base voltage that makes it non-conductive. As a result, the output side of the amplifier 12 is almost equal to the insertion of a one-stage high-pass filter, and its attenuation characteristic is, for example, 6 dB / oct.

When the reverberation increases, the deviation of the detected level ratios of both microphones 11 and 21 from the predetermined value increases, and the transistor Tr of the second-stage filter 134 has a roll-off frequency of 100 Hz. Control signal forming circuit 2 so as to have a stable impedance
9 is controlled by the control signal from the amplifier 9. As a result, the amplifier 1
The output signal of 2 is almost equal to the insertion of a two-stage high-pass filter, and its attenuation characteristic is, for example, 12 dB / oct.

Further, when the picked-up sound field is close to the diffused sound field and there is a lot of reverberation, the deviation of the level ratio from the default value becomes larger, and the transistor Tr of the third-stage filter 135 rolls at 100 Hz. The control signal forming circuit 29 has an impedance that provides an off frequency.
Controlled by the control signal from
Is almost equal to the insertion of a three-stage high-pass filter, for example, its attenuation characteristic is
It is set to 18 dB / oct.

In this way, in this example, since the steepness of the attenuation characteristic of the high-pass filter is changed according to the picked-up sound field, the picked-up quality is always good even in the picked-up sound field with a lot of reverberation. So that it is automatically controlled.

Also in this example, as in the example of FIG. 2, a low-pass filter or a band-pass filter is inserted between the amplifier 12 and the level detection circuit 25 and between the amplifier 22 and the level comparison circuit 23. The high-pass filter 13 may be controlled based on the level ratio of only the low frequency range of the microphone output.

In this example, the filters 133 ...
Transistor T when 135 is inserted into the output signal
It is also possible to control the roll-off frequency also by changing the base voltage of r.

In the above example, there is no one way to construct the directional microphone 11. That is, the microphone unit having the directivity may be used as it is, or the target directivity may be obtained by combining a plurality of microphone units.

For example, by arranging two omnidirectional microphone units slightly apart from each other, unidirectional or bidirectional output can be obtained.

An example in which a unidirectional microphone is realized by using two omnidirectional microphone units will be described with reference to FIGS. 4 and 5. As shown in FIG. 4, in this example, the omnidirectional microphone units 30 and 31 are
It is placed a small distance d away. Then, as shown in FIG. 5, the output audio signal of one microphone unit 30 is passed through an amplifier (not shown) to the subtraction circuit 32.
Is supplied to. The output audio signal of the other microphone unit 31 is similarly supplied to the subtraction circuit 32 via an amplifier and filter circuit 33 (not shown). The filter circuit 33 includes a resistor 34 and a capacitor 35 in this example.
Composed of and.

In this case, assuming that the resistance value of the resistor 34 is R1 and the capacity of the capacitor 35 is C1, the resistance value R1 and the capacity are set so that C1R1 = d / c (where c is the speed of sound). C1 has been selected.

In this example, the output of the subtraction circuit 32 is output to the output terminal 37 via the frequency characteristic correction circuit 36 such as an integrator for flattening the frequency characteristic. As will be described later, the frequency characteristic correction circuit 36 is provided as needed and need not be provided.

The operation of the microphone of this example will be described. As shown in FIG. 4, the sound source is in the direction of an angle θ with respect to the arrangement direction of the two microphone units 30, 31, and the two microphone units 30, 3 are arranged.
1 is incident on each unit 30, 31
If the outputs of P0 and P1 are P0 and P1, then the output P1 is P1 = P0 ε- ^{jω (d / c) cos θ} . Note that ω is the angular frequency.

Since the output of the microphone unit 31 is supplied to the subtraction circuit 32 through the filter circuit 33, the output signal Pa of the subtraction circuit 32 is as shown in the following expression 6.

[0073]

[Equation 6] In Expression 6, A represents the filter function of the filter circuit 33, and ω · d / c << 1.

If the following expression 7 is satisfied in expression 6, the output Pa exhibits unidirectionality.

[0075]

[Equation 7] That is, when the equation (7) is satisfied, the equation (6) is Pa = P0.jω (d / c) (1 + cos θ), and the angle θ has unidirectionality.

In the case of the above example, the filter function A of the filter circuit 33 is expressed by A = 1 / (1 + jωC1R1), and C1R1 = d / c, so that Since A = 1 / (1 + jωd / c), it is clear from the equation (7) that the microphone of the embodiment of FIG. 4 has unidirectionality. However, the frequency characteristic of this microphone rises to the right (the higher the frequency, the larger the response). In this example, the frequency characteristic correction circuit 36 is provided to flatly correct the upward rising characteristic.

In the example of FIG. 5, the filter circuit 3
3, subtraction circuit 32, and further frequency characteristic correction circuit 36
Can also be realized by a digital filter or a processing program (software). For example, the filter circuit 33 can be configured with a digital filter including an adder 41, a delay circuit 42, and a feedback amplifier 43 of the transfer function A as shown in FIG.

Similarly, omnidirectional or bidirectional output can be freely obtained by the two unidirectional microphone units arranged in the back or facing each other.

FIG. 7 is a block diagram of an embodiment in that case. This example is an example of a 2-channel stereo microphone, 51R is for collecting right channel sound, and 51R
L is a unidirectional microphone for collecting left channel sound. These are microphones having a flat sound pressure-frequency characteristic in a free sound field, and they are provided close to each other, and the microphone 51R has a left pointing channel so that the main pointing axis faces the right channel sound direction. The voices are arranged so that the directions of the voices are directed to the main directivity axis.

The output signal of the microphone 51R is supplied to the high pass filter 53R via the amplifier 52R, and the signal from the high pass filter 53R is led to the output terminal 54R. Also, the output signal of the microphone 51L is passed through the amplifier 52L and passed through the high pass filter 5.
The signal from the high-pass filter 53L is supplied to the output terminal 54L. These high pass filters 53R and 53L are the high pass filters 1 described above.
Similar to 3, the roll-off frequency and the attenuation characteristic can be made variable.

The output signals of the amplifiers 52R and 52L are supplied to the adder circuit 55 and added. Then, the addition circuit 55 provides an output signal equivalent to that picked up by a microphone whose directional characteristic is almost omnidirectional. The output signal of the adding circuit 55 is supplied to the level detecting circuit 56, the level is detected, and the detected output is supplied to the level comparing circuit 57.

Further, the output signals of the amplifiers 52R and 52L are respectively supplied to the subtraction circuit 58 to be subtracted. Then, from the subtraction circuit 55, an output signal equivalent to that picked up by a bidirectional microphone having a directivity characteristic of approximately 8 is obtained. The output signal of the subtraction circuit 58 is supplied to the level detection circuit 59, the level is detected, and the detection output is supplied to the level comparison circuit 57.

Then, the level comparison circuit 57 supplies an output signal having a level ratio of both input signals to the control signal formation circuit 50. Then, the control signal from the control signal forming circuit 50 causes the filters 53R and 53L to have the roll-off frequency or the attenuation characteristic in the same manner as in the above-described example.
It is variably controlled according to the deviation between the level ratio and a predetermined value. As a result, a stereo signal with good sound quality can be obtained.

In the example of FIG. 7 as well, a low-pass filter or a band-pass filter is provided on the input side of the level detection circuits 56 and 59, and the sound pickup sound field is discriminated from the level ratio of only the low frequency component. By doing so, the determination becomes more reliable.

Although the above example is an example in which an analog filter is used as the high pass filter, for example, a first-order or second-order IIR digital high pass filter can be used. In that case, the roll-off frequency and the attenuation characteristic can be controlled by setting the filter coefficient of the IIR digital filter according to the output level ratio between the microphones 11 and 21.

In this case, it is necessary to convert the output signal of the amplifier 12 into a digital signal and supply it to the digital high-pass filter so that the filter output is returned to the analog signal. Further, the means for detecting and comparing the level ratio and the circuit configuration of the control signal forming circuit can also be realized by a digital circuit configuration.

As described above, since the amount of bass sound of the microphone output can be adjusted appropriately depending on the degree of the sound of the sound collecting sound field, the sound collecting quality of voice and the like can be made good and not unclear. Can be output to.

Moreover, even if the sound condition of the sound collecting sound field changes due to the movement of the microphone or the like, the adjustment is automatically performed, so that sound collecting quality with no discomfort can be obtained. Also, it is convenient for the user to concentrate on other work without worrying about the adjustment.

Next, an embodiment of a microphone device to which the adaptive noise reduction device is applied will be described below.

In this type of microphone device, as described above, when the picked-up sound field is diffuse and the reverberation is increased, or when the background noise level becomes equal to or higher than the level of desired voice, adaptive noise reduction is performed. Since the rejection rate of unnecessary noise is limited due to the nature of the system, similar to the example of the microphone device described above, the ratio of the output level of the omnidirectional microphone to the output level of the directional microphone produces a sound pickup field. The environment is determined and the adaptive processing is controlled according to the result of the determination.

Similar to the above-mentioned example, the directional microphone output may be unidirectional, bidirectional, or super directional, and if it is not omnidirectional, all of them are provided in order to solve the problems of the present invention. Available.

An embodiment of this type of microphone device will be described below with reference to FIGS. 8 and 9.

In FIG. 8, reference numeral 61 is a main input microphone for picking up a desired voice, and 71 is a reference input microphone for picking up unnecessary voice and ambient noise in a direction to be removed as noise. In this example, as in the case of FIG. 11 described above, the arrival direction of the desired voice is mainly the direction from the upper side to the lower side in the figure (hereinafter referred to as the front direction), as indicated by the arrow AR in FIG. This is an example of realizing a microphone device that does not pick up sound from a direction opposite to the direction (hereinafter referred to as a back direction) as noise.

In the case of this example, the main input microphone 61 is composed of an omnidirectional microphone as shown in FIG. 11A. In this example, the microphone 61 has a flat sound pressure-frequency characteristic in a free sound field, and in this example, it is also used as an omnidirectional microphone for discriminating a sound recording field.

On the other hand, as shown in FIG. 11B, the reference input microphone 71 is a unidirectional microphone having no sensitivity in the desired voice arrival direction and only in the back direction. The microphone 71 has a sound pressure-frequency characteristic equal to that of the microphone 61 in a free sound field. In this example, the sound pressure-frequency characteristic is flat. In this example, the directional microphone for discriminating the sound collecting sound field is used. Also used as. The microphone 71 can be configured by using a plurality of omnidirectional microphone units as described above.

Then, the sound signal S picked up by the main input microphone 61 and converted into an electric signal is obtained.
The signal 61 is supplied to the A / D converter 63 via the amplifier 62, converted into a digital signal, and supplied to the subtraction circuit 65 via the delay circuit 64.

Further, a voice signal S71 obtained by collecting the sound by the reference input microphone 71 and converting it into an electric signal.
Is supplied to the A / D converter 73 via the amplifier 72, converted into a digital signal, and supplied to the adaptive filter circuit 74.

In this embodiment, the adaptive filter circuit 74
Is composed of an FIR filter type adaptive linear combiner 300 and an arithmetic circuit for adaptively controlling the linear combiner 300, in this example, an LMS arithmetic circuit 320, and an A / D converter 73. Is supplied to the subtraction circuit 65 via the linear combiner 300. The adaptive filter circuit 74 is a DS including a microprocessor.
It can be configured by P (digital signal processor).

The output signal of the subtraction circuit 65 is fed back to the arithmetic circuit 320 of the adaptive filter circuit 74, and D
The signal is converted back into an analog signal by the / A converter 66 and is led to the output terminal 67.

It should be noted that the D / A converter 66 is omitted and the output signal of the subtraction circuit 65 remains as a digital signal at the output terminal 67.
May be derived. In addition, the delay circuit 64 is
This is for compensating for a time delay such as a propagation time in the adaptive filter circuit 74 or a time delay required for calculation for adaptive processing.

The adaptive filter circuit 74 controls so that the reference input voice signal approximates the noise component contained in the main input voice signal. As a result, if it is assumed that the desired voice and noise in the voice picked up by the main input microphone 61 are uncorrelated, the subtraction circuit 65 extracts the reference input microphone 71 from the voice signal of the main input microphone 61. The voice signal (noise) is subtracted and removed, and only the desired voice signal is obtained from the subtraction circuit 65.

That is, in the configuration of this embodiment, the output audio signal of the main input microphone 61 is supplied as the main input, and the output audio signal of the reference input microphone 71 is supplied as the noise as the reference input. It has a reduction system configuration.

FIG. 9 shows an example in which an algorithm of the so-called LMS (Least Mean Square) method is used as an adaptive algorithm for one embodiment of the adaptive filter circuit 74.

As shown in FIG. 8, an FIR filter type adaptive linear combiner 300 is used as the adaptive filter circuit 74 in this example. This is a plurality of delay circuits DL1, each having a delay time Z ⁻¹ of a unit sampling time.
DL2, ... DLm (m is a positive integer) and input signal n1
And weighting circuits MX0, MX1, for multiplying the output signals of the delay circuits DL1, DL2 ,.
MX2, ..., MXm, and an adder circuit 310 for adding the outputs of the weighting circuits MX0 to MXm. The output of the adder circuit 310 is y.

The weighting coefficients supplied to the weighting circuits MX0 to MXm are formed from the residual signal e from the subtraction circuit 65 in the LMS operation circuit 320 composed of, for example, a microcomputer. The algorithm executed by the LMS arithmetic circuit 320 is as follows.

Now, the input vector Xk at time k is
As shown in FIG. 9, X _k = [x _0k x _1k x _2k ... x _mk ] ^T , the output is y _k , and the weighting coefficient is w _jk (j = 0,1,2, ... m). Then, the relationship of input and output is as shown in the following Equation 8.

[0107]

[Equation 8] Becomes

Then, the weight vector W _{k at} time _k
Is defined as W _k = [w _0k w _1k w _2k ... w _mk ] ^T , the input / output relationship is given by y _k = X _k ^T · W _k . If the desired response is d _k , the error e _{k from the} output is expressed as follows. e _k = d _k −y _k = d _k −X _k ^T · W _{k In the} LMS method, the weight vector is updated by the formula W _{k + 1} = W _k +2 μ · e _k · X _k . Here, μ is a gain factor (step gain) that determines the speed and stability of adaptation.

In this way, noise, unnecessary voice signals from the back side in this example, are selectively removed at the output terminal 67, and as a result, the desired voice signal remains and is output. In other words, the super-directivity is substantially realized. In addition, even if a voice is emitted in the vicinity from a direction other than the front direction, the sound is appropriately suppressed, so that a better sound collection quality can be obtained.

By the way, in order to reduce the noise in the main input by using the reference input by the adaptive processing as described above, the desired speech and the reference noise need to be uncorrelated. For this reason, conventionally, in this type of adaptive noise reduction system, the microphone for reference input has been devised so that it does not pick up the desired voice as the reference input, or it is installed as close as possible to the noise source. Measures are taken such as keeping it away from the input microphone. However, this makes the system large and difficult to move.

On the other hand, in the case of this example, the desired voice and the noise are distinguished according to the arrival direction of the voice. The main input microphone 61 has a directional characteristic (including omnidirectionality) capable of picking up a voice from a desired voice arrival direction, and also has a reference input microphone 71.
Is a structure in which the desired voice does not have sensitivity in the incoming direction or has a low sensitivity directivity, the desired voice in the voice picked up by the main input microphone 61 and the reference input microphone 71 are picked up. It is designed to be uncorrelated with noise.

Therefore, in the case of this example, it is only necessary to consider the directivity of the main input microphone and the reference input microphone, and it is possible to dispose both microphones close to each other, as compared with the conventional microphone system. Can be small. Then, the noise signal is well removed from the main input, and as a result, a directional microphone device having low sensitivity or no sensitivity in the direction of arrival of the noise input can be easily realized.

However, as described above, if the picked-up sound field is diffuse and reverberant, or if the background noise level is equal to or higher than that of the desired voice, adaptive noise reduction processing is expected. However, it is difficult to obtain good sound collection quality.

In order to improve this, a level ratio detecting circuit 81 is provided in this embodiment. The level ratio detection circuit 81 includes amplifiers 62 and 72 and A / D converters 63 and 7.
The audio signals S61 and S71 from the microphones 61 and 71 are supplied via 3 and the levels of the audio signals from the microphones 61 and 71 are compared. The output signal of the level ratio detection circuit 81 is supplied to the arithmetic circuit 320 of the adaptive filter circuit 74 as a control signal.

The arithmetic circuit 320 compares the predetermined level ratio determined by the directivity of the microphones 61 and 71 with the level ratio detected by the level ratio detection circuit 81,
Limit the adaptive processing according to the deviation from the default value. Specifically, the coefficient update of the adaptive filter is controlled.

That is, if the deviation between the level ratio (S71 / S61) of the reference input signal to the main input signal and the predetermined value is not more than the predetermined value, the adaptive filter circuit 74 performs the above-mentioned normal adaptive processing. To do so.

The input signal level ratio (S71 / S
When the deviation between 61) and the predetermined value becomes larger than the predetermined value, the output of the level ratio detection circuit 81 limits the update of the weighting coefficient of the adaptive filter circuit 74. The following three types of methods can be considered as the method of this limitation.

(1) Step gain μ = 0. That is, the coefficient is not updated and is fixed to the immediately preceding coefficient. (2) The step gain μ is made smaller than a normal value to reduce the update amount. For example, the step gain is set to 1/10. (3) The step gain μ is set to 0 and the filter coefficient value is set to a default value prepared in advance.

The methods (1) and (2) can be removed from the situation where unnecessary noise is not eliminated by stopping or suppressing the coefficient updating in the adaptive filter circuit 74. In the method (3), for example, a coefficient that forms a hypercardioid as a directivity is set in the adaptive filter to suppress the unnecessary signal on average. In this case, unnecessary noise is eliminated as the microphone having the sharpest directivity among the primary sound pressure gradient type microphones called hypercardioid.

As described above, according to the example of FIG. 8, it is possible to estimate from the output level ratio of the microphones 61 and 71 the degree of reverberation of the picked-up sound field or the relative magnitude of the background noise level. The directivity of the output can be controlled. For example, when collecting sound in a place with a high background noise level, it is possible to escape from the situation in which unnecessary noise is rarely excluded and continue sound collection with a directivity that minimizes the directivity coefficient, at least as with hypercardioid. .

In the example of FIG. 8 as well, the microphone 6
By determining the situation of the picked-up sound field from the level ratio of only the low frequency range of the output audio signals 1 and 71, the detection accuracy of the level ratio is improved, and the situation of the picked-up sound field is more accurately discriminated and appropriate. Can be obtained.

Further, the main input microphone 61 may be a unidirectional microphone arranged so that its directional axis is oriented in the desired voice arrival direction. In this case, the omnidirectional microphone for determining the sound pickup field can obtain the omnidirectional microphone output by adding the outputs of the main input microphone 61 and the reference input microphone 71. .

In the above example, the main input microphone 61 and the reference input microphone 71 are also used for discriminating the sound collecting sound field. The omnidirectional microphone and the directional microphone may be separately provided. The directional microphone can be composed of a plurality of omnidirectional microphone units as described above.

[0124]

As described above, according to the present invention, the amount of bass can be adjusted appropriately depending on the amount of sound in the picked-up sound field. You can output without.

Moreover, even if the sound condition of the sound collecting sound field changes due to the movement of the microphone or the like, the adjustment is automatically performed, so that sound collecting quality with no discomfort can be obtained. Therefore, it is convenient for the user to concentrate on other work without worrying about the adjustment.

Further, according to the present invention, the coefficient update of the adaptive noise reduction device can be controlled appropriately depending on the sound level of the picked-up sound field, so that the picked-up sound field has a lot of reverberation.
Even when the background noise level is relatively large and the effect of the original adaptive noise reduction device cannot be expected, the directivity of the system is appropriately controlled to improve the sound collection quality of voice and the like. It can be output without being obscured.

[Brief description of drawings]

FIG. 1 is a block diagram showing a configuration of an embodiment of a microphone device according to the present invention.

FIG. 2 is a block diagram showing the configuration of another embodiment of the microphone device according to the present invention.

FIG. 3 is a block diagram showing the configuration of another embodiment of the microphone device according to the present invention.

FIG. 4 is a diagram for explaining an example of configuring a microphone with a plurality of microphone units.

FIG. 5 is a diagram showing an example in which a microphone is configured by a plurality of microphone units.

FIG. 6 is a diagram showing another configuration example of a part of the example of FIG.

FIG. 7 is a block diagram showing the configuration of another embodiment of the microphone device according to the present invention.

FIG. 8 is a block diagram showing the configuration of another embodiment of the microphone device according to the present invention.

9 is a diagram showing an example of the adaptive filter circuit of the example of FIG.

FIG. 10 is a diagram for explaining an example of a conventional microphone device.

FIG. 11 is a diagram for explaining an arrangement example of microphones.

FIG. 12 is a diagram showing an equivalent circuit of a condenser microphone.

[Explanation of symbols]

 11 Directional Microphone 13 High Pass Filter 14 Output Terminal 21 Omnidirectional Microphone 24 Level Comparison Circuit 25 Control Signal Forming Circuit 27, 28 Low Pass Filter 61 Main Input Microphone 65 Subtraction Circuit 71 Reference Input Microphone 74 Adaptive Filter Circuit 81 Level ratio detection circuit

─────────────────────────────────────────────────── ─── Continuation of the front page (51) Int.Cl. ⁵ Identification number Reference number within the agency FI Technical indication location H04R 1/40 320 Z (72) Inventor Kaoru Gyotoku 6-735 Kitashinagawa, Shinagawa-ku, Tokyo Inside Sony Corporation

Claims (7)

[Claims] 1. A directional first microphone, an omnidirectional second microphone, comparing means for comparing output levels of the first and second microphones, and a first microphone of the first microphone. A high-pass filter provided for the output, wherein the device output signal is taken out from the high-pass filter, and the roll-off frequency of the high-pass filter is based on the output of the comparison means. A microphone device that is controlled so as to change according to the level ratio of the outputs of the first and second microphones. 2. The microphone device according to claim 1, wherein the steepness of the attenuation characteristic of the high pass filter is changed according to the level ratio of the outputs of the first and second microphones. Microphone device characterized by. 3. The microphone device according to claim 1, wherein two omnidirectional microphone units arranged close to each other are used to form the first and second microphones. Microphone device characterized by. 4. The microphone device according to claim 1 or 2, wherein two unidirectional microphone units arranged close to each other are formed to form the first and second microphones. A microphone device characterized by being used. 5. A first microphone for picking up a desired voice, a directional second microphone having low sensitivity in the direction of arrival of the desired voice, and a voice signal from the second microphone are supplied. Adaptive filter means, subtracting means for subtracting the output signal of the adaptive filter means from the voice signal of the first microphone, and means for adjusting the adaptive filter means so that the output power of the subtracting means is minimized. A directional third microphone, an omnidirectional fourth microphone, a level ratio detecting means for detecting a level ratio of the output audio signals of the third and fourth microphones, and the level ratio detection A means for limiting the adaptive operation of the adaptive filter means based on the deviation between the level ratio detected by the means and a predetermined value. 6. The microphone device according to claim 5, wherein the third and fourth microphone outputs are obtained from the first and second microphone outputs. 7. The microphone device according to claim 5, wherein when the deviation between the output level ratio of the third and fourth microphones and a predetermined value exceeds a preset value. A microphone device, wherein the coefficient of an adaptive filter is set to a predetermined value according to its size.