JP3064685B2 - Microphone device - Google Patents

Description

DETAILED DESCRIPTION OF THE INVENTION

[0001]

BACKGROUND OF THE INVENTION 1. Field of the Invention The present invention relates to a microphone device.

[0002]

2. Description of the Related Art For example, in a camera-integrated VTR, while photographing a subject, sounds around the subject are simultaneously recorded. For this recording, a microphone device having directivity that mainly collects only sound from the direction of the subject, that is, the front direction of the camera, is generally used.

As an example of this type of microphone device,
A high-order sound pressure gradient type, so-called gun microphone, is known. As shown in FIG. 10, the gun microphone has a configuration in which a pipe portion 2 is coaxially disposed in front of a diaphragm 1 (right side in the figure), and a number of through holes 3 are provided in a side wall of the pipe portion 2. You.

As shown in FIG. 10A, for a sound wave arriving from the front of the microphone, the path length from reaching the diaphragm 1 through each through hole 3 is equal to the path length from the tip of the pipe portion 2. It becomes the same, and the sound waves of each path are added in phase and reinforce each other. On the other hand, as shown in FIGS. 10B and 10C, for sound waves arriving from the side or rear of the microphone, the path lengths from the through holes 3 to the diaphragm 1 are different, and the sound waves of each path are added at different phases. Being weakened as a whole.

In this manner, a microphone having a single directivity, which has high sensitivity to sounds from the front and low sensitivity to sounds from the sides and the rear, can be obtained.

[0006]

However, the above-mentioned microphone has a drawback that it requires a long pipe portion and becomes large. Further, even with the unidirectionality as described above, there is a problem that unnecessary sound emitted near the camera is loudly collected even if the sound is incident from the side or the rear.

[0007] In view of the above, the present invention is compact, and collects only sound incident from a desired direction.
A first object is to provide a microphone device capable of appropriately suppressing other sounds. It is a second object of the present invention to prevent distortion of incident sound from a desired direction.

[0008]

In order to solve the above-mentioned problems, a microphone device according to the present invention is provided with a first microphone 11 for picking up a desired sound when corresponding to reference numerals in the embodiments described later. A second microphone 21 having directivity with low sensitivity in the direction of arrival of a desired sound, adaptive filter means 24 to which a sound signal from the second microphone is supplied, and an output signal of the adaptive filter means being output to a first microphone. Subtracting means 15 for subtracting from the audio signal of
Level ratio detecting means 51 for detecting the level ratio of the audio signals of the first and second microphones, and adjusting the adaptive filter means so as to minimize the output power of the subtracting means. , The operation of the adaptive filter means is limited >>.

[0009]

According to this configuration, the audio signal from the second microphone 21 having low sensitivity in the direction of arrival of the desired voice is transmitted to the first microphone.
Has little correlation with the desired sound from the microphone 11 of
The microphone device of the present invention has a configuration of an adaptive noise reduction system, and when the output power of the subtraction unit is minimized, the audio signal of the second microphone 21 is removed from the audio signal from the first microphone 11. , Only the desired sound is output.

When the audio signal of the second microphone 21 is lower than a predetermined level, the adaptive processing is restricted. Thus, even if the input level of the second microphone is low, it is possible to prevent the incident sound in the desired direction from being distorted. That is, the microphone device of the present invention has sensitivity only in the direction of arrival of a desired sound,
Appropriately suppresses sound entering from other directions.

[0011]

DETAILED DESCRIPTION OF THE PREFERRED EMBODIMENTS An embodiment of a microphone device according to the present invention will be described below with reference to FIGS.

In FIG. 1, reference numeral 11 denotes a main input microphone for picking up desired sound, and reference numeral 21 denotes a reference input microphone for picking up unnecessary sound and ambient noise in a direction to be removed as noise. In this example, the direction in which the desired sound arrives is mainly a direction from above to below in the figure (hereinafter referred to as a front direction) as shown by an arrow AR in FIG. This is an example of realizing a microphone device that does not pick up the sound from the above as noise.

In the case of this example, the main input microphone 11 is constituted by a non-directional microphone as shown in FIG. On the other hand, as shown in FIG. 2, the reference input microphone 21 is a unidirectional microphone having no sensitivity in the direction of arrival of the desired sound and having sensitivity only in the back direction.

The audio signal collected by the main input microphone 11 and converted into an electric signal is supplied to an A / D converter 13 via an amplifier 12 and converted into a digital signal. The signal is supplied to the subtraction circuit 15 via the delay circuit 14. The audio signal collected by the reference input microphone 21 and converted into an electric signal is supplied to an A / D converter 23 via an amplifier 22 to be converted into a digital signal, and is converted into a digital signal by an adaptive filter circuit. 24.

In this embodiment, the adaptive filter circuit 24
Is composed of an FIR filter type adaptive linear combiner 300 as shown in FIG. 3 and an arithmetic circuit for adaptively controlling the linear combiner 300, in this example, an LMS arithmetic circuit 320. Is supplied to the subtraction circuit 15 via the linear combiner 300. The adaptive filter circuit 24 includes a microprocessor DS
It can be constituted by P (digital signal processor).

The output signal of the subtraction circuit 15 is fed back to the arithmetic circuit 320 of the adaptive filter circuit 24,
The signal is converted back to an analog signal by the -A converter 16 and output to the output terminal 17.

Further, in this embodiment, a level ratio detection circuit 51 is provided, and the microphones 11 and 21 are connected to the level ratio detection circuit 51 via amplifiers 12 and 22 and A / D converters 13 and 23. The audio signals S11 and S21 are supplied and the levels of the main and reference input audio signals are compared. The output signal of the level ratio detection circuit 51 is supplied to the arithmetic circuit 320 of the adaptive filter circuit 24 as a control signal.

The DA converter 16 is omitted, and the output signal of the subtraction circuit 15 is output as a digital signal at the output terminal 17.
May be derived. In addition, the delay circuit 14
This is for compensating for a time delay such as a propagation time in the adaptive filter circuit 24 and a time delay required for an operation for adaptive processing.

The adaptive filter circuit 24 controls the reference input audio signal to approximate the noise component contained in the main input audio signal, as described later. This allows
Assuming that the desired voice and the noise in the voice collected by the main input microphone 11 are uncorrelated, the subtraction circuit 15 uses the voice signal of the reference input microphone 21 ( Noise) is subtracted and removed, and only the desired audio signal is obtained from the subtraction circuit 15. In other words, the configuration of this embodiment is different from that of the adaptive noise reduction system in which the output audio signal of the main input microphone 11 is supplied as the main input and the output audio signal of the reference input microphone 21 is supplied as the noise as the reference input. It has a configuration. First, the basic operation of this system will be described. In this case, the main input audio signal from the A / D converter 13 is obtained by adding the desired audio signal s from the front direction of the arrow AR and the noise n0 from the rear direction considered to be uncorrelated with this. is there. On the other hand, assuming that the reference input audio signal from the A / D converter 23 is n1, as is apparent from the above description, this reference input audio signal n1
Is uncorrelated with the desired speech signal, but is correlated with the noise n0. The adaptive filter circuit 24 filters the reference input audio signal n1 and outputs a signal y. The adaptive algorithm is a subtraction signal (residual signal) output from the subtraction circuit 15.
Signal) works to minimize e.

Now, assuming that s, n0, n1, and y are statistically stationary and the average value is 0, the output is e = s + n0-y. The expected value of squaring 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 30 converges, the adaptive filter circuit 30 is adjusted so that E [e ² ] is minimized. At this time, since the E [s ^2] is not affected, and ^{Emin [e 2] = E [} s 2] + Emin [(n0 -y) 2]. That is, by minimizing E [e ² ], E [(n 0 −y) ² ] is minimized, and the output y of the adaptive filter circuit 30 becomes an estimated amount of the noise n 0. The expected value of the output from the subtraction circuit 15 is only the desired signal. That is, adjusting the adaptive filter circuit 30 to minimize the total output power is equivalent to the fact that the subtraction output e becomes the least square estimation value of the desired audio signal s.

FIG. 3 shows an embodiment of the adaptive filter circuit 30 in which a so-called LMS (least mean square) algorithm is used as an adaptive algorithm.

As shown in FIG. 3, in this example, an FIR filter type adaptive linear combiner 300 is used. this is,
A plurality of delay circuits DL1, DL2,... DLm (m is a positive integer) each having a delay time Z ^-1 of a unit sampling time, an input signal n1, and each of the delay circuits DL1, DL
.., MXm for multiplying the output signal of the DLm by the weighting coefficient, and an adder circuit 310 for adding the outputs of the weighting circuits MX0 to MXm. The output of the adder 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 15 by an LMS operation circuit 320 composed of, for example, a microcomputer. The algorithm executed by the LMS operation circuit 320 is as follows.

Now, the input vector Xk at time k is
As shown in FIG. 3, X _k = [x _0k x _1k x _2k ... X _mk ] ^T , the output is y _k , and the weighting factor is w _jk (j = 0, 1, 2,... M). Then, the relationship between the input and output is as shown in the following equation 1.

[0025]

(Equation 1) Becomes

Then, the weight vector W _{k at} time _k
Is defined as W _k = [w _{0 k} w _{1 k} w _{2 k} ... W _mk ] ^T , the input / output relationship is given by y _k = X _k ^T · W _k . Assuming that the desired response is d _k , the error e _{k from the} output is expressed as follows. In the e _k = d _k -y _k = d _k -X _k ^T · W _k LMS method, the updating of the weight vector is performed by the following equation: 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 manner, noise, in this case, an unnecessary audio signal from the back direction, is selectively removed from the output terminal 17, and as a result, the desired audio signal remains output. In other words, the super directivity is substantially realized. In addition, even if sound is emitted in the vicinity from a direction other than the front direction, the sound is appropriately suppressed, so that better sound pickup 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 voice and the reference noise need to be uncorrelated as described above. For this reason, conventionally, in this type of adaptive noise reduction system, the reference input microphone has been devised so as to prevent the desired sound from being picked up as a reference input, or installed as close as possible to the noise source. Measures such as keeping the microphone away from the input microphone have been taken. However, this makes the system bulky and difficult to move.

On the other hand, in the case of the present invention, desired sound and noise are distinguished according to the direction of arrival of the sound. And
The main input microphone 11 has a directional characteristic (including omnidirectionality) capable of picking up sound from a desired sound arrival direction, and has a reference input microphone 2.
Reference numeral 1 denotes a configuration in which the desired sound arrival direction has no sensitivity or has a low sensitivity directivity, and the desired sound in the sound collected by the main input microphone 11 and the reference sound collected by the reference input microphone 21 are included. Noise and no correlation.

Therefore, in the case of the present invention, it is only necessary to consider the directivity of the main input microphone and the reference input microphone, and it is possible to arrange both microphones close to each other. And can be small.

According to the configuration of the present invention, the noise signal is satisfactorily removed from the main input. As a result, a directional microphone device with low sensitivity or no sensitivity in the direction of arrival of the noise input can be easily realized. it can. FIG.
It is a figure for showing what checked the effect in the case of this example experimentally.

That is, as shown in FIG. 2, this experimental apparatus uses the main input microphone 11 and the reference input microphone 2 with the desired sound arrival direction as the direction of the arrow AR.
1 are arranged before and after the main input microphone 11 along the desired sound arrival direction. Then, for example, a sine wave signal of 1 kHz is made to arrive from the direction of arrow AR as a desired sound, and a sine wave of 600 Hz is made to arrive as noise from the direction of the back and, for example, 30 ° so that the sound is collected. Do.

In this example, the sensitivity of the omnidirectional main input microphone 11 is 0 dB, the reference input microphone 21 is -20 dB for a sound from the front, the sensitivity in the back direction is 0 dB, and the sensitivity of the reference input microphone 21 is 30 dB from the back. The sensitivity to voice input from the direction of ° is -0.7 dB.

At this time, the input waveform of the main input microphone 11 is a combination of a 1 kHz sine wave and a 600 Hz sine wave as shown in FIG. The sound waveform is as shown in FIG. 4B, which approximates the 1 kHz sine wave of the ideal output waveform shown in FIG. 4C. As a result, the basic effects of the microphone device according to the present invention were confirmed.

By the way, in order to reduce noise in the main input by using the reference input by the adaptive processing as described above, the reference input is uncorrelated with the desired voice and at a certain level as described above. It is assumed that it exists. However, for example, in a situation where a performance by a musical instrument player in the front direction is collected, in a concert or the like,
In order to pick up sound in an environment where noise does not occur around,
It is conceivable that the level of the reference input is considerably low, or that only the desired sound is generated, and there is substantially no reference input. In such a case, it becomes equivalent to the input of a very small residual noise to the reference input, and the characteristic of the adaptive filter circuit 24 greatly deviates from the originally desirable one.
As described above, since the noise reduction system works to minimize its output power, the desired audio signal will also suffer some distortion.

In order to improve this, in this embodiment, a level ratio detection circuit 51 is provided to compare the levels of the main input voice signal and the reference input voice signal (noise), and compare the level of the main input voice signal with that of the main input voice signal. When the level of is reduced, the adaptive processing is restricted.

That is, if the level ratio (S21 / S11) of the reference input signal to the main input signal is equal to or more than a predetermined value (that is, if an unnecessary signal is incident at a certain level or more), the adaptive filter circuit 24 A normal adaptive process such as is performed.

The input signal level ratio (S21 / S
When 11) is smaller than the predetermined value, the output of the level ratio detection circuit 51 restricts the updating of the weighting coefficient of the adaptive filter circuit 24. There are the following three methods for this restriction. (1) Set the step gain μ = 0.

(2) The step gain μ is made smaller than a normal value. For example, 1/10.

(3) Set the step gain μ = 0 and set the filter coefficient value to a default value prepared in advance.

The methods (1) and (2) stop or suppress the updating of the coefficients in the adaptive filter circuit 24 so that the coefficients at the time when the unnecessary signal is appropriate are determined by the level ratio (S21 / S21).
11) is maintained and operated until an appropriate value is obtained. With this method, it is possible to suppress a large change in the adaptive filter due to the desired signal and the residual noise.

In the method (3), for example, a coefficient that forms a single directivity as the directivity is set in the adaptive filter to perform average suppression of unnecessary signals.

By the above-described limitation of the adaptive operation, it is possible to prevent the desired sound signal from being distorted when the level of the reference input is lowered, and to restore the level of the reference input after the restoration.
Normal adaptive processing is performed in the adaptive filter circuit 24, so that noise in the main input audio signal can be appropriately reduced.

Next, another embodiment of the microphone device according to the present invention will be described with reference to FIGS. FIG. 5 shows a configuration of a main part of another embodiment of the present invention.

In FIG. 5, reference numerals 30 and 31 denote omnidirectional microphone units, which are arranged at a distance d as shown in FIG. The output signals of the two microphone units 33, 31 are supplied to the non-inverting input terminals of the operational amplifiers 32, 33, respectively, as a pair of subtraction circuits, and the inverting input terminals of the operational amplifier 32 output the output of the microphone unit 31. The signal is supplied via a filter circuit 34, and the output signal of the microphone unit 31 is supplied to an inverting input terminal of the operational amplifier 33 via a filter circuit 35.

In this example, the filter circuit 34 is configured as a low-pass type by a resistor 34r and a capacitor 34c, and the other filter circuit 35 is similarly configured. When the resistances of the resistors 34r and 35r are R1 and the capacitances of the capacitors 34c and 35c are C1, the resistance value R1 and the capacitance are set so that C1.R1 = d / c (where c is the speed of sound). C1 has been selected.

In this example, the operational amplifiers 32, 3
The output signal of No. 3 is led to a pair of output terminals 38 and 39 via frequency characteristic correction circuits 36 and 37 such as integrators for flattening the frequency characteristics. The signals obtained at the output terminals 38 and 39 are equivalently regarded as a main input audio signal and a reference input audio signal (noise) from a pair of unidirectional microphones as described later. 1 are supplied to the A / D converters 13 and 23, respectively. Other configurations are the same as those in the example of FIG.

In the embodiment shown in FIG. 5, as described below, the operational amplifiers 32 and 33 convert one output signal of a pair of omnidirectional microphone units 30 and 31 from the other through a filter circuit 34 or 35. Are subtracted from each other to realize a pair of microphones each having a single directivity.

As shown in FIG. 6, there is a sound source in the direction of an angle of θ with respect to the arrangement direction of the two microphone units 30 and 31, and these two microphone units 3
When it is assumed that light is incident on 0, 31, each unit 3
Assuming that the outputs of 0 and 31 are P0 and P1, the output P1 is P1 = ^{P0ε-jω (d / c) cos θ} . Here, ω is an angular frequency.

Since the output of the microphone unit 31 is supplied to one subtraction circuit 32 through the filter circuit 34, the output signal Pa of the subtraction circuit 32 is as shown in the following equation (2).

[0051]

(Equation 2) In Equation 2, A represents a filter function of the filter circuit 34, and ω · d / c << 1.

In Equation 2, if this filter function A satisfies the following Equation 3, the output Pa indicates unidirectionality.

[0053]

(Equation 3) That is, when the filter function A satisfies the equation (3),
The above equation (2) becomes Pa = P0.A.j.omega. (D / c) (1 + cos .theta.), And monotonically changes with respect to the angle .theta. From the maximum value at .theta. = 0 to the minimum value at .theta. = 180. That is, the output signal Pa of the subtraction circuit 32 becomes unidirectional with respect to the angle θ.

By the way, in the case of the above example, the filter function A of the filter circuit 34 is represented by A = 1 / (1 + jωC 1 · R 1), and C 1 · R 1 = d / c. A = 1 / (1 + jωd / c), which is equal to Equation 3. From the above description, it is clear that in the embodiment of FIG. 5, the output signal of one subtraction circuit 32 exhibits unidirectionality.

Similarly, the output signal of the other subtraction circuit 33 also exhibits unidirectionality, but as apparent from FIG. 5, the subtraction circuit 33 has a microphone unit in a reverse relationship to the subtraction circuit 32. Since the output signals P0 and P1 of the subtractors 30 and 31 are supplied, the output signals of the two subtraction circuits 32 and 33 are
It exhibits opposite unidirectionalities.

However, the output signal Pa of the above-described subtraction circuit is
Includes the frequency factor ω in the coefficient related to the angle θ,
The frequency characteristic rises to the right (response increases as the frequency increases)
Characteristic. In this example, the frequency characteristic correction circuit 36,
37 is provided to correct the upward-sloping characteristic flat.

Thus, in the embodiment of FIG. 5, while using a pair of non-directional microphone units 30 and 31,
Equivalently, as shown in FIG. 8, a pair of unidirectional microphones arranged in opposite directions are realized.

In the example of FIG. 5, the filter circuit 3
4, 35, the subtraction circuits 32 and 33, and the frequency characteristic correction circuits 36 and 37 can also be realized by a digital filter or a processing program (software). For example, as shown in FIG. 7, the filter circuits 34 and 35 can be configured by a digital filter including an adder 41, a delay circuit 42, and a feedback amplifier 43 having a transfer function A.

In the case of the embodiment of FIG. 5, equivalently, the main input microphone 11 and the reference input microphone 21 of the example of FIG. 1 are unidirectional, as shown in FIG. The main input microphone 11 is located at the front, and the microphone 11 is arranged before and after along the desired sound arrival direction. In the case of the example of FIG. 5, the main input microphone 11
Is oriented with the highest sensitivity towards the front,
The reference input microphone 21 is arranged, for example, with the direction of the highest sensitivity facing the back. That is, the reference input microphone 21 has low sensitivity in the direction of arrival of the desired sound, and has high sensitivity in the back direction, which is the direction of arrival of noise in this example.

Therefore, also in this example, the desired voice in the main input voice and the noise collected as the reference input become uncorrelated, and the noise is selectively reduced by the adaptive processing as described above. Thus, it is possible to realize a microphone device that obtains only desired sound as an output.

If the level ratio of the reference input signal to the main input signal (S21 / S11) is equal to or greater than a predetermined value, the adaptive filter circuit 24 (see FIG. 1) performs the normal adaptive processing as described above. In this example, the direction of arrival of the noise is
If the range is approximately 90 degrees from the back direction, the sensitivity of the main input microphone 11 is low in this range, so that the noise level during the main input is low. Therefore,
The main input microphone 11 itself has a noise reduction effect.

Also in this embodiment, when the input signal level ratio (S21 / S11) falls below a predetermined value, the above-described methods (1), (2), and (3) for limiting the adaptive operation can be performed by: That is, a countermeasure by suspending or suppressing the updating of the weighting coefficient, and the weighting circuits MX0 to M0 of the adaptive filter circuit 24.
The method of setting a default (predetermined) weighting factor for Xm prevents the desired voice in the main input from being distorted. In the case of this example, when the method (3) is adopted, a weighting coefficient such that the transfer characteristic of the adaptive filter circuit 24 becomes [0.5] is loaded, as shown in FIG. A pair of unidirectional microphones 11 and 21 arranged in opposite directions form hypercardioid directivity.

As a result, even when the level of the reference input is lowered, the noise can be reduced on average while avoiding the distortion of the desired voice signal. Adaptive processing can be performed to appropriately reduce noise in the primary input speech signal.

Incidentally, the microphone of the hyper cardioid has a directivity index of -6 dB, and a directivity index of -5 dB.
It has sharper directivity than the unidirectional microphone of dB, and it can be said that the directivity is sharpest except for the high-order sound pressure gradient type as described above. Here, the directivity index (directivity index)
index) is the direct logarithm of directivity efficiency. The directivity is defined by the following equation (4).

[0065]

(Equation 4) Here, D (Ω) represents the ratio of the output voltage to the incident wave at the angle Ω to that of Ω = 0, and dΩ represents a small solid angle in the direction of the angle Ω.

The directional efficiency represents the ratio of the energy response of a directional microphone to the energy response of a omnidirectional microphone having the same front sensitivity to sound randomly incident from all directions with equal probability. The directivity index, which is a common logarithm of the directivity, has a larger negative value as the directivity becomes sharper, and is particularly meaningful in a diffuse sound field.

In this embodiment, a level ratio detection circuit 51 is provided to provide a pair of unidirectional microphones 11 and 12.
Although the level of the output signal is compared, when the desired sound from the front is mainly incident, the level ratio of both signals is (S21 / S11) <1, and if expressed in logarithm, it is minus. Become. On the other hand, when the unwanted sound mainly enters from the back direction, the level ratio of both signals is (S21 / S11)>
It becomes 1 and it becomes a plus if expressed in logarithm. Therefore,
By monitoring the signal level ratio (S21 / S11), it is possible to easily estimate whether the main sound wave is from the front or from the back.

In each of the above-described embodiments, as shown in FIGS. 2 and 8, a microphone unit having directivity with high sensitivity in the rear direction is used for the reference input. When it is desired to limit the direction of arrival to the vicinity of the 90-degree direction or to increase the sensitivity of the reference input microphone in the 90-degree direction, as shown in FIG. Character directivity)
It is said. In the case of FIG. 9, the main input microphone 1
1 is a unidirectional one arranged so as to have the highest sensitivity in the desired voice direction as in the example of FIG.
As in the example of FIG. 1, it may be omnidirectional.

In each of the above embodiments, the microphone device according to the present invention has been described as an example of a sound-collecting microphone device of a camera-integrated VTR. However, the present invention is not limited to this example. Of course,
Professional video cameras, measurement microphone devices, etc.
Applicable to all microphone devices.

Further, in each of the above embodiments, the adaptive filter circuit is constituted by a digital circuit, so that the entire circuit is constituted by a digital circuit. However, the adaptive filter circuit is constituted by an analog circuit. Alternatively, only the adaptive filter circuit portion may have a digital configuration. Further, when the level of the reference input is reduced, various countermeasures based on weighting coefficients of the adaptive filter circuit may be used in combination.

[0071]

As described above, according to the present invention, the first microphone for picking up the desired sound,
A second microphone having directivity having low sensitivity in the direction of arrival of the desired sound is arranged close to the microphone, and an output signal of the second microphone is supplied to a subtraction unit via an adaptive filter circuit. Subtracting the output signal of the adaptive filter means from the output signal of the first microphone, adjusting the adaptive filter means so as to minimize the output power of the subtraction means, and detecting the level ratio of the output signals of both microphones. Since the operation of the adaptive filter means is restricted based on the output of the level ratio detecting means, it is possible to collect only a sound incident from a desired direction with a small size and to appropriately suppress other sounds. A microphone device can be realized.

[Brief description of the 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 diagram illustrating an example of directivity of first and second microphones.

FIG. 3 is a diagram illustrating an example of the adaptive filter circuit of the example of FIG. 1;

FIG. 4 is a diagram for explaining the operation of the microphone device according to the present invention.

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

FIG. 6 is a diagram illustrating an example in which a microphone is configured by a plurality of microphone units.

FIG. 7 is a diagram illustrating another configuration example of a part of the example of FIG. 5;

FIG. 8 is a diagram illustrating another example of the directivity of the first and second microphones.

FIG. 9 is a diagram showing still another example of the directivity of the first and second microphones.

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

[Explanation of symbols]

 11, 21 microphone 15 subtraction circuit 24 adaptive filter circuit 51 level ratio detection circuit

Continuation of the front page (72) Inventor Kaoru Gyokudori Sony Corporation, 6-7-35 Kita Shinagawa, Shinagawa-ku, Tokyo (56) References JP-A-61-150497 (JP, A) JP-A-64-39195 (JP, A) JP-A-4-181898 (JP, A) (58) Fields investigated (Int. Cl. ⁷ , DB name) H04R 3/00 320 H03H 21/00 H04R 1/40 320

Claims (4)

(57) [Claims] 1. A first microphone for recording a desired sound, a second microphone having a low sensitivity in an arrival direction of the desired sound, and a sound signal from the second microphone are supplied. Adaptive filter means; subtraction means for subtracting the output signal of the adaptive filter means from the audio signal of the first microphone; means for adjusting the adaptive filter means so that the output power of the subtraction means is minimized. A level ratio detecting means for detecting a level ratio of the audio signals of the first and second microphones; and a means for restricting an adaptive operation of the adaptive filter means based on an output of the level ratio detecting means. Microphone device. 2. The method according to claim 1, wherein the adaptive operation of said adaptive filter means is restricted.
2. The microphone device according to claim 1, wherein the updating of the weighting coefficient of the adaptive filter means is suspended. 3. An adaptive operation of said adaptive filter means is restricted.
Microphone device according to claim 1, wherein the step gain is possible to reduce for updating the weighting coefficient of the adaptive filter means. 4. The adaptive operation of the adaptive filter means is restricted.
2. The microphone device according to claim 1, wherein the weighting coefficient of said adaptive filter means is switched to a predetermined value.