JP2004187283A - Microphone unit and reproducing apparatus - Google Patents

Abstract

<P>PROBLEM TO BE SOLVED: To provide a microphone unit capable of stably operating even under a plurality of noises in the practical environment and of realizing a high S/N. <P>SOLUTION: A signal generating part generates a main signal and a noise reference signal. A judgement part judges whether or not a level ratio is greater than a predetermined value. An adaptive filter part generates a signal indicating a signal component of a target sound included in the noise reference signal generated by the signal generating part and learns a filter coefficient only when it is judged by the judgement part that the level ratio is greater than the predetermined value. A subtraction part subtracts the signal generated by the adaptive filter part. A noise suppressing part uses the main signal and the noise reference signal after subtraction by the subtraction part to suppress a signal component of noise contained in the main signal. <P>COPYRIGHT: (C)2004,JPO&NCIPI

Description

  The present invention relates to a microphone device and a sound reproducing device, and more specifically, to a microphone device and a sound reproducing device for detecting a sound arriving from a predetermined direction while suppressing noise.

The configuration of a conventional microphone device will be described with reference to FIGS.
FIG. 24 is a diagram illustrating a configuration of the microphone device of the first conventional example. 24, the microphone device includes a first microphone unit 1010, a second microphone unit 1020, a signal adding unit 1030, a first signal subtracting unit 1031, a signal amplifying unit 1050, and an adaptive filter unit 1060. , And a second signal subtraction unit 1062. Each of the microphone units 1010 and 1020 is arranged so as to face the front direction (left direction in FIG. 24). Signal adding section 1030 adds the signal output from first microphone unit 1010 and the signal output from second microphone unit 1020. First signal subtraction section 1031 subtracts a signal output from second microphone unit 1020 from a signal output from first microphone unit 1010. Signal amplifying section 1050 multiplies the signal output from signal adding section 1030 by １ ／. Adaptive filter section 1060 receives the signal output from first signal subtraction section 1031 as an input, and outputs a signal that has been filtered by an adaptive filter. Second signal subtraction section 1062 subtracts the signal output from adaptive filter section 1060 from the signal output from signal amplification section 1050. The output from the second signal subtraction unit 1062 is the output of the microphone device. Adaptive filter section 1060 learns filter coefficients based on the signal output from second signal subtraction section 1062 and the signal output from first signal subtraction section 1031.

  Next, the operation of the microphone device of Conventional Example 1 will be described. When detecting sound coming from the front direction, each of the microphone units 1010 and 1020 outputs substantially the same signal. When detecting a sound arriving from a direction other than the front direction, each of the microphone units 1010 and 1020 outputs a signal having a different phase. Output signals from the microphone units 1010 and 1020 are added by the signal adding unit 1030. The level of the added signal is normalized by the signal amplifying unit 1050, that is, the amplitude is doubled. As described above, a main signal having a sound component arriving from the front direction can be obtained. On the other hand, according to the output from the first signal subtraction unit 1031, the directivity main axis is oriented in the 90-degree direction with respect to the front direction, and the front direction becomes the directivity blind spot (that is, the front direction is the minimum directivity). (Directivity direction). That is, the signal output from the first signal subtraction unit 1031 is a noise reference signal that does not include a sound component arriving from the front. Adaptive filter section 1060 realizes adaptive directivity by using the main signal output from signal amplification section 1050 and the noise reference signal output from first signal subtraction section 1031. That is, a directional blind spot is automatically formed for a noise source in one direction coming from a direction other than the front direction.

  FIG. 25 is a diagram illustrating a configuration of a microphone device of Conventional Example 2. In FIG. 25, the microphone device includes a first microphone unit 1010, a second microphone unit 1020, a first adaptive filter unit 1040, a first signal delay unit 1041, a first signal subtraction unit 1042, , A second adaptive filter unit 1060, a second signal delay unit 1061, and a second signal subtraction unit 1062.

  First adaptive filter section 1040 receives an output signal from second microphone unit 1020 as an input, and outputs a result of filtering by the adaptive filter. First signal delay section 1041 delays a signal output from first microphone unit 1010. First signal subtraction section 1042 subtracts the signal output from first adaptive filter section 1040 from the signal output from first signal delay section 1041. First adaptive filter section 1040 learns filter coefficients based on the signal output from first signal subtraction section 1042 and the signal output from second microphone unit 1020. Second signal delay section 1061 delays the signal output from first signal delay section 1041. Second adaptive filter section 1060 receives a signal output from first signal subtraction section 1042 as an input, and outputs a result of filtering by the adaptive filter. The second signal subtraction unit 1062 subtracts the signal output from the second adaptive filter unit from the signal output from the second signal delay unit 1061 to obtain an output of the microphone device. Second adaptive filter section 1060 learns filter coefficients based on the signal output from second signal subtraction section 1062 and the signal output from the first signal subtraction section.

  Hereinafter, the operation of the microphone device of Conventional Example 2 will be described. The first adaptive filter unit 1040, the first signal delay unit 1041, and the first signal subtraction unit 1042 in the second conventional example perform a cancel operation on sound waves arriving at the microphone units 1010 and 1020. . That is, the signal output from first signal subtraction section 1042 serves as a noise reference signal for second adaptive filter section 1060. The signal output from the first signal subtraction unit 1042 is the same target signal as the signal output from the first signal subtraction unit 1031 illustrated in FIG. However, the conventional example 1 has a fixed directivity, whereas the conventional example 2 is different in that the directivity can be changed by using an adaptive filter.

  FIG. 26 is a diagram illustrating a configuration of a microphone device of Conventional Example 3. The microphone device shown in FIG. 26 includes a first unidirectional microphone unit 1011, a second unidirectional microphone unit 1012, a first FFT unit 1070, a second FFT unit 1080, and two inputs. A type spectrum subtraction unit 1090 and a speech recognition unit 2000 are provided.

  In FIG. 26, the first unidirectional microphone unit 1011 is arranged so that the main axis of directivity faces the front. The second unidirectional microphone unit 1012 is arranged so that the principal axis of directivity faces the back. First FFT section 1070 obtains a frequency spectrum by using a signal output from first unidirectional microphone unit 1011 as an input. Second FFT section 1080 receives a signal output from second unidirectional microphone unit 1012 as an input, and obtains a frequency spectrum. Two-input type spectrum subtraction section 1090 receives signals output from each of FFT sections 1070 and 1080 as inputs, and obtains a signal spectrum derived by second FFT section 1080 from a signal spectrum derived by first FFT section 1070. Is subtracted in the power spectrum region to output the spectrum of the target signal. Speech recognition section 2000 performs speech recognition using the spectrum of the target signal output from two-input type spectrum subtraction section 1090 as an input.

Hereinafter, the operation of the microphone device of Conventional Example 3 will be described. In Conventional Example 3, the first unidirectional microphone unit 1011 has a directional characteristic for collecting a target sound in the front direction. The second unidirectional microphone unit 1012 has a directional characteristic for mainly collecting noise. As a result, the main signal m1 is obtained from the first unidirectional microphone unit 1011 and the noise reference signal m2 is obtained from the second unidirectional microphone unit 1012. In each of FFT sections 1070 and 1080, the spectra of main signal m1 and noise reference signal m2 are obtained. In two-input type spectral subtraction section 1090, the power spectrum of the signal component is estimated by subtracting the power spectrum of the noise reference signal from the power spectrum of the main signal. In the one-input type spectral subtraction method, the noise spectrum is estimated on the assumption that the noise is stationary in a time section where the target sound does not arrive. Therefore, in the one-input type spectral subtraction method, only stationary noise suppression can be performed. On the other hand, according to the configuration of Conventional Example 3 employing the two-input type spectral subtraction method, since the second unidirectional microphone unit 1012 can always obtain the spectrum of the noise reference signal, It is possible to suppress noise. As described above, according to the microphone device of the third conventional example, the speech recognition rate of the subsequent speech recognition unit 2000 can be improved by suppressing not only stationary noise but also non-stationary noise. The device shown in FIG. 26 is used for speech recognition. Here, it is also possible to obtain a microphone device by returning the spectrum to a time signal by performing IFFT at the final stage and converting the spectrum to a waveform signal with frame overlap.
Japanese Patent No. 3084833 Bernard Widow, "ADAPTIVE SIGNAL PROCESSIN", Prentice Hall, 1985, p. 414, 419, 423 Nakadai, Kanmura, Nakatsu, "Speech Recognition in Vehicles Using Noise Removal Method with Two Inputs", IEICE Technical Report, 1989, SP89-81, pp. 146-64. 41-48

  In the configuration of the above-described conventional example 1, a large noise suppression effect can be obtained in an environment where noise comes from a certain direction. However, the device of Conventional Example 1 cannot deal with noise coming from a plurality of directions. Therefore, in an actual noise environment in which noise sources exist in various directions at the same time, the configuration of the conventional example 1 can obtain only a noise suppression effect equivalent to the performance of a conventionally used unidirectional microphone device. I can't.

  In the configuration of Conventional Example 2, a noise reference signal is obtained by using the first adaptive filter. Here, in order to operate the first adaptive filter stably in the real environment, it is necessary to learn the first adaptive filter only when the voice from the speaker is sufficiently larger than the surrounding noise. Therefore, in the configuration of the second conventional example, the noise suppression effect cannot be obtained until the convergence of the filter is completed. Also, it is difficult to converge the filter in a noise environment. Further, like the first conventional example, the configuration of the second conventional example cannot cope with a plurality of noise sources. Further, since the device of the second conventional example is invented for the purpose of suppressing wind noise having no correlation between unit signals, the direction of the target sound cannot be limited. That is, the loudest sound among the incoming sounds is the target sound, and it is not possible to pick up sound in a specific direction while enhancing it.

  The configuration of Conventional Example 3 is a system in which a main signal and a noise reference signal are converted into a spectrum, and noise is suppressed using a spectrum subtraction method in a power spectrum. This method is a method capable of simultaneously suppressing noise even when there are noise sources in a plurality of directions. However, this method has a problem that even if the target sound is slightly mixed in the noise reference signal, a serious problem occurs in the sound quality of the processed sound or the target sound itself is canceled. Further, in an actual sound field, even if the directional blind spot of the unidirectional microphone unit is directed to the target sound direction, it is conceivable that the reflected wave wraps around and mixes. Further, in a normal microphone unit, since the blind spot of the directivity is not an infinite attenuation but an attenuation of about 10 to 15 db, a direct wave of a target sound may not be completely removed and may be mixed into a noise reference signal. is there. In addition, in the case of the spectrum subtraction method, a processing delay due to the frame processing occurs, so that there is a problem that the method cannot be used for applications such as simultaneous communication and public address.

  Further, the above-mentioned conventional example focuses on suppression of additive noise which is another noise different from the target sound. In the above conventional example, multiplicative noise that arrives after the target sound reaches a reflecting surface such as a wall, a desk, or a floor cannot be removed. Therefore, the frequency characteristics of the target sound may be distorted due to the influence of reflection or the like in a sound field where the microphone device is actually used. Therefore, particularly in applications such as speech recognition, it has not been possible to solve the problem of erroneous recognition that mismatch occurs in matching at the time of recognition.

  Therefore, an object of the present invention is to provide a microphone device that can operate stably even under a plurality of noises in an actual use environment and can realize a high S / N.

  It is another object of the present invention to provide a microphone device that suppresses both multiplicative noise caused by a reflected wave of a target sound and additive noise caused by the noise.

  It is another object of the present invention to generate a main signal and a noise reference signal used in a process for suppressing noise by a simple method.

  In order to achieve the above object, the present invention employs the following configurations. That is, the first invention is a microphone device that detects a target sound arriving from a target sound direction. The microphone device includes a signal generation unit, a determination unit, an adaptive filter unit, a subtraction unit, and a noise suppression unit. The signal generation unit indicates a main signal indicating a result of detection with sensitivity to the target sound direction and a result of detecting a sound arriving from a direction other than the target sound direction with higher sensitivity than the target sound. And a noise reference signal. The determining unit determines whether a level ratio indicating a ratio of a signal level of the main signal to a signal level of the noise reference signal generated by the signal generating unit is larger than a predetermined value. The adaptive filter unit generates a signal indicating a signal component of a target sound included in the noise reference signal generated by the signal generation unit by filtering the main signal generated by the signal generation unit with the adaptive filter, and performs determination. Only when the unit determines that the level ratio is larger than the predetermined value, learning of the filter coefficient is performed. The subtractor subtracts the signal generated by the adaptive filter from the noise reference signal generated by the signal generator. The noise suppression unit suppresses a signal component of noise included in the main signal using the main signal and the noise reference signal after the subtraction by the subtraction unit.

  The “main signal indicating the result of detection with sensitivity to the target sound direction” refers to not only the signal itself output from the microphone unit but also the signal detected by the microphone unit that has been subjected to predetermined processing. This includes signals obtained as a result. That is, the main signal may be a signal itself output from a microphone unit whose directivity main axis is directed in the target sound direction, or may be a microphone unit (which may be non-directional, The signal may be a signal obtained by processing a signal output from the directional main axis. Similarly, the “noise reference signal indicating the result of detecting a sound arriving from another direction with higher sensitivity than the target sound” may be the signal itself output from the microphone unit or the microphone unit. May be a signal obtained by processing the signal output from the.

  A second invention is a microphone device for detecting a target sound arriving from a target sound direction. The microphone device includes a signal generation unit, a determination unit, an adaptive filter unit, a subtraction unit, a reflection information calculation unit, and a reflection correction unit. The signal generation unit indicates a main signal indicating a result of detection with sensitivity to the target sound direction and a result of detecting a sound arriving from a direction other than the target sound direction with higher sensitivity than the target sound. And a noise reference signal. The determining unit determines whether a level ratio indicating a ratio of a signal level of the main signal to a signal level of the noise reference signal generated by the signal generating unit is larger than a predetermined value. The adaptive filter unit generates a signal indicating a signal component of a target sound included in the noise reference signal generated by the signal generation unit by filtering the main signal generated by the signal generation unit with the adaptive filter, and performs determination. Only when the unit determines that the level ratio is larger than the predetermined value, learning of the filter coefficient is performed. The subtractor subtracts the signal generated by the adaptive filter from the noise reference signal generated by the signal generator. The reflection information calculation unit calculates information on the arrival time difference between the direct wave and the reflected wave of the target sound based on the filter coefficient of the adaptive filter unit. The reflection correction unit corrects a frequency characteristic distortion generated in the main signal due to the reflected wave of the target sound based on the information calculated by the reflection information calculation unit.

  In the third invention, the signal generation unit includes a first microphone unit and a second microphone unit. The first microphone unit is arranged such that the directivity main axis is directed to the target sound direction. The second microphone unit is arranged such that the blind spot direction of the directivity is directed to the target sound direction.

  Further, in the fourth invention, the microphone device further includes a signal delay unit. The signal delay unit is provided between the output terminal of the noise reference signal in the signal generation unit and the subtraction unit, and delays the noise reference signal so as to satisfy the convergence condition of the adaptive filter of the adaptive filter unit.

  In the fifth aspect, the predetermined value can be changed.

  In the sixth aspect, the signal generation unit includes a first microphone unit, a second microphone unit, a delay unit, an amplification unit, a first subtraction unit, and a second subtraction unit. . The second microphone unit has the same characteristics as the first microphone unit. The delay unit delays a signal output from the first microphone unit by a predetermined delay amount and outputs the signal. The amplification unit amplifies the signal output from the delay unit. The first subtraction unit generates a main signal by subtracting a signal amplified by the amplification unit from a signal output from the second microphone unit. The second subtraction unit generates a noise reference signal by subtracting the signal output from the delay unit from the signal output from the second microphone unit. Further, a straight line connecting the position of the first microphone unit and the position of the second microphone unit is orthogonal to the straight line directed to the target sound direction. The predetermined delay amount is set such that the noise reference signal contains more sound components arriving from directions other than the target sound direction than components of the target sound. The amplification factor in the amplifying unit is set such that a difference occurs in the sensitivity of the target sound between the main signal and the noise reference signal.

  In the seventh aspect, the microphone device further includes a setting unit that changes a predetermined delay amount set in the delay unit.

  In the eighth invention, the signal generation unit includes a first microphone unit, a second microphone unit, and a synthesis unit. The second microphone unit has the same characteristics as the first microphone unit. The combining unit generates a main signal based on each signal output from the first and second microphone units so as to have sensitivity in a target sound direction, and minimizes the sensitivity in the target sound direction. To generate a noise signal component.

In the ninth aspect, the signal generation unit includes a first microphone unit, a second microphone unit, a signal addition unit, and a signal subtraction unit.
The second microphone unit is arranged with the directivity main axis directed in a direction different from that of the first microphone unit. The signal adding unit generates a main signal by adding a signal output from the first microphone unit and a signal output from the second microphone unit. The signal subtraction unit generates a noise reference signal by subtracting one of the signal output from the first microphone unit and the signal output from the second microphone unit from the other.

  In the tenth aspect, the signal generation unit includes a first microphone unit, a second microphone unit, a stereo signal generation unit, an inverse synthesis unit, and a synthesis unit. The second microphone unit has the same characteristics as the first microphone unit. The stereo signal generation unit generates a stereo signal including a right channel signal and a left channel signal based on the first and second microphone units. The inverse synthesizing unit generates each signal output from each microphone unit based on the stereo signal. The synthesizing unit, based on each signal generated by the inverse synthesizing unit, converts a main signal indicating a result of detection with sensitivity to the target sound direction and a sound arriving from a direction other than the target sound direction. A noise reference signal indicating a result of detection with higher sensitivity than the target sound is generated.

  In the eleventh invention, the signal generation unit includes a first microphone unit, a second microphone unit, a stereo signal generation unit, a signal addition unit, and a signal subtraction unit. The second microphone unit has the same characteristics as the first microphone unit. The stereo signal generation unit generates a stereo signal including a right channel signal and a left channel signal based on the first and second microphone units. The signal adding unit generates a main signal by adding the right channel signal and the left channel signal of the stereo signal. The signal subtraction unit generates a noise reference signal by subtracting one of the right channel signal and the left channel signal of the stereo signal from the other.

  Further, in the twelfth aspect, the microphone device further includes a reflection information calculation unit and a reflection correction unit. The reflection information calculation unit calculates information on the arrival time difference between the direct wave and the reflected wave of the target sound based on the filter coefficient of the adaptive filter unit. The reflection correction unit corrects a frequency characteristic distortion generated in the main signal due to the reflected wave of the target sound based on the information calculated by the reflection information calculation unit. The noise suppression unit suppresses a signal component of noise included in the main signal by using the main signal corrected by the reflection correction unit and the noise reference signal after subtraction by the subtraction unit.

  Further, in the thirteenth invention, the noise suppression unit includes a noise suppression filter coefficient calculation unit and a time-varying coefficient filter unit. The noise suppression filter coefficient calculation unit calculates a filter coefficient of a noise suppression filter for suppressing a noise signal component from the main signal based on the main signal and the noise reference signal after the subtraction by the subtraction unit. The time-varying coefficient filter unit performs filtering on the main signal by the noise suppression filter, reflecting the filter coefficient calculated by the noise suppression filter coefficient calculation unit.

  Further, in the fourteenth invention, the noise suppression filter coefficient calculation unit includes a first frequency analysis unit, a second frequency analysis unit, a power spectrum ratio calculation unit, and a multiplication unit, and a coefficient calculation unit. In. The first frequency analysis unit calculates a power spectrum of the main signal. The second frequency analysis unit calculates a power spectrum of the noise reference signal after the subtraction by the subtraction unit. The power spectrum ratio calculation unit calculates the power spectrum calculated by the first frequency analysis unit and the power spectrum calculated by the second frequency analysis unit only when the level ratio is determined to be smaller than the predetermined value by the determination unit. The time average of the power spectrum ratio with the power spectrum is calculated. The multiplication unit multiplies the time average of the power spectrum ratio calculated by the power spectrum ratio calculation unit and the power spectrum calculated by the second frequency analysis unit. The coefficient calculation unit calculates a filter coefficient of the noise suppression filter based on the power spectrum calculated by the first frequency analysis unit and a result of the multiplication by the multiplication unit.

  A fifteenth invention is a microphone device for detecting a target sound arriving from a target sound direction. The microphone device includes a first microphone unit, a second microphone unit, a signal adding unit, a signal subtracting unit, and a noise suppressing unit. The second microphone unit is arranged with the directivity main axis directed in a direction different from that of the first microphone unit. The signal adding unit generates a main signal by adding a signal output from the first microphone unit and a signal output from the second microphone unit. The signal subtraction unit generates a noise reference signal by subtracting one of the signal output from the first microphone unit and the signal output from the second microphone unit from the other. The noise suppression unit suppresses a signal component of noise included in the main signal using the main signal and the noise reference signal.

  A sixteenth invention is a microphone device for detecting a target sound arriving from a target sound direction. The microphone device includes a first microphone unit, a second microphone unit, a stereo signal generation unit, an inverse synthesis unit, a synthesis unit, and a noise suppression unit. The second microphone unit has the same characteristics as the first microphone unit. The stereo signal generation unit generates a stereo signal including a right channel signal and a left channel signal based on the first and second microphone units. The inverse synthesizing unit generates each signal output from each microphone unit based on the stereo signal. The synthesizing unit, based on each signal generated by the inverse synthesizing unit, converts a main signal indicating a result of detection with sensitivity to the target sound direction and a sound arriving from a direction other than the target sound direction. A noise reference signal indicating a result of detection with higher sensitivity than the target sound is generated. The noise suppression unit suppresses a signal component of noise included in the main signal using the main signal and the noise reference signal.

  Further, a seventeenth invention is a microphone device for detecting a target sound arriving from a target sound direction. The microphone device includes a first microphone unit, a second microphone unit, a stereo signal generation unit, a signal addition unit, a signal subtraction unit, and a noise suppression unit. The second microphone unit has the same characteristics as the first microphone unit. The stereo signal generation unit generates a stereo signal including a right channel signal and a left channel signal based on the first and second microphone units. The signal adding unit generates a main signal by adding the right channel signal and the left channel signal of the stereo signal. The signal subtraction unit generates a noise reference signal by subtracting one of the right channel signal and the left channel signal of the stereo signal from the other. The noise suppression unit suppresses a signal component of noise included in the main signal using the main signal and the noise reference signal.

  Further, an eighteenth aspect includes a voice recording unit, a signal generation unit, a determination unit, an adaptive filter unit, a subtraction unit, a noise suppression unit, and a reproduction unit. The audio recording unit records audio signals of at least two types of channels. The signal generation unit includes a main signal indicating a result of detection with sensitivity to the target sound direction based on the audio signal recorded in the recording unit, and a sound arriving from a direction other than the target sound direction. And a noise reference signal that indicates the result of detecting with a higher sensitivity than the target sound. The determining unit determines whether a level ratio indicating a ratio of a signal level of the main signal to a signal level of the noise reference signal generated by the signal generating unit is larger than a predetermined value. The adaptive filter unit generates a signal indicating a signal component of a target sound included in the noise reference signal generated by the signal generation unit by filtering the main signal generated by the signal generation unit with the adaptive filter, and performs determination. Only when the unit determines that the level ratio is larger than the predetermined value, learning of the filter coefficient is performed. The subtractor subtracts the signal generated by the adaptive filter from the noise reference signal generated by the signal generator. The noise suppression unit suppresses a signal component of noise included in the main signal using the main signal and the noise reference signal after the subtraction by the subtraction unit. The reproduction unit reproduces the main signal whose noise signal component has been suppressed by the noise suppression unit.

  In the nineteenth aspect, the microphone device includes a video recording unit that records a video signal related to the audio signal recorded in the audio recording unit, and a video reproduction unit that reproduces a video signal recorded in the video recording unit. And a direction receiving unit that receives an input of a direction in which sound should be emphasized from the user. At this time, the signal generation unit generates the main signal and the noise reference signal with the direction received by the direction reception unit as the target sound direction.

  According to the first aspect, the signal component of the target sound included in the noise reference signal is removed from the noise reference signal, and thereafter, noise suppression processing is performed based on the main signal and the noise reference signal. Therefore, since noise suppression processing can be performed using an ideal noise reference signal, high S / N can be realized. According to the first aspect, all sounds other than the target sound can be suppressed as noise. Therefore, it is possible to cope with not only noise in one direction but also noise in all directions.

  Further, according to the second aspect, since the influence of the reflected wave on the main signal can be corrected, a microphone device having a stable sensitivity versus frequency characteristic without being influenced by a sound field around the microphone device is realized. can do. In addition, since there is no change in sound quality due to the reflector, the effect of improving the recognition rate is particularly great for speech recognition.

  According to the third aspect, the main signal and the noise reference signal can be easily generated. Further, since the two microphone units can be arranged close to each other until they come into contact with each other, the microphone device can be downsized.

  Further, according to the fifth aspect, it is possible to control how many left and right sound pickup areas of the microphone device can be picked up around the target sound direction. Therefore, it is possible to set the sound collection angle width according to the purpose, or to make the sound collection angle width variable like a zoom microphone.

  According to the sixth aspect, it is possible to obtain a directivity pattern in which the sensitivity characteristics of the main signal and the noise reference signal substantially match in directions other than the target sound direction. Therefore, the consistency in the subsequent noise suppression processing is improved, and the quality of the processed speech is improved.

  Further, according to the seventh aspect, the sound collection direction can be controlled by changing the delay time.

  According to the ninth aspect, for example, a main signal and a noise reference signal can be obtained by using a signal output from a one-point stereo microphone.

  According to the tenth and eleventh aspects, a main signal and a noise reference signal can be obtained using a stereo signal.

  Further, according to the twelfth aspect, it is possible to simultaneously suppress both the noise that is additive noise and the reflected wave that is multiplicative noise. Therefore, it is possible to realize a microphone having a high S / N ratio and always flat frequency characteristics without being affected by the sound field.

  According to the fifteenth, sixteenth, and seventeenth aspects, the main signal and the noise reference signal used in the noise suppression processing can be generated by a simple method. For example, it can be generated by adding simple processing to the configuration of a conventional stereo microphone.

(Embodiment 1)
First, a microphone device according to Embodiment 1 of the present invention will be described with reference to FIGS. FIG. 1 is a block diagram showing a configuration of the microphone device according to the first embodiment. In FIG. 1, the microphone device includes a first microphone unit 1, a second microphone unit 2, a determination unit 10, an adaptive filter unit 20, a signal subtraction unit 30, a noise suppression filter coefficient calculation unit 40, And a time-varying coefficient filter unit 50.

  In FIG. 1, the first microphone unit 1 is a unidirectional microphone unit. The directivity main axis of the first microphone unit 1 is directed in the front direction. The second microphone unit 2 is a bidirectional microphone unit. The directivity main axis of the second microphone unit 2 is directed in a direction perpendicular to the front direction. Note that the microphone device detects sound coming from a desired direction. Hereinafter, a sound to be detected is called a target sound, and the desired direction is called a target sound direction. In the first embodiment, the front direction is the target sound direction.

  The determination unit 10 uses the signal m1 output from the first microphone unit 1 and the signal m2 output from the second microphone unit 2 as input signals, and determines whether or not a target sound arrives according to a level ratio between the input signals. Is determined. The adaptive filter unit 20 outputs a signal obtained by filtering the signal m1 with a filter coefficient. The signal subtraction unit 30 subtracts the signal output from the adaptive filter unit 20 from the signal m2.

  The noise suppression filter coefficient calculation unit 40 inputs the signal m1 as a main signal, and inputs the signal m3 output from the first signal subtraction unit 30 as a noise reference signal. The noise suppression filter coefficient calculation unit 40 calculates a filter coefficient indicating a filter characteristic for noise suppression using the main signal and the noise reference signal. The calculated filter coefficients are output to the time-varying coefficient filter unit 50. The time-varying coefficient filter unit 50 receives the signal m1. Then, the input signal is filtered and output according to the filter coefficient calculated by the noise suppression filter coefficient calculation unit 40.

  The operation of the microphone device configured as described above will be described. In the following description, unless otherwise specified, the direction from which the target sound arrives is the front direction.

  In FIG. 1, a first microphone unit 1 is arranged close to a second microphone unit 2. By arranging the microphone units 1 and 2 close to each other, the second microphone unit 2 can collect sound (ie, noise) other than the target sound at substantially the same position as the first microphone unit 1. it can. The microphone device according to the first embodiment realizes high S / N sound collection by suppressing noise mixed in the first microphone unit 1 by the time-varying coefficient filter unit 50. At that time, the signal m2 is used as a noise reference signal. Therefore, it is ideal that each of the microphone units 1 and 2 collects sound in the same sound field. That is, it is desirable that the microphone units 1 and 2 be arranged in contact with each other, provided that the microphone units 1 and 2 do not affect the formation of the directivity of the microphone units 1 and 2. Therefore, in the first embodiment, the microphone units 1 and 2 are arranged close to each other.

  Further, in the first embodiment, a noise suppression processing method using a time-varying coefficient filter is adopted in the subsequent stage of the microphone device (the noise suppression filter coefficient calculating unit 40 and the time-varying coefficient filter unit 50). When the target sound is mixed into the second microphone unit 2, due to the nature of the method, an adverse effect such as distortion or a level drop occurs in the processed voice. Therefore, when this method is used, the problem is how to remove the target sound from being mixed into the noise reference signal. Therefore, in the first embodiment, in order to minimize mixing of the target sound into the noise reference signal, the second microphone unit 2 is configured so that the blind spot of the directivity faces the front. The reason why the bidirectional microphone unit is used as the second microphone unit 2 is that the bidirectional microphone unit has a manufacturing variation with respect to characteristics such as a blind spot direction and a sensitivity attenuation amount. This is because there is a feature that the number is smaller than that of the microphone unit.

  By configuring the second microphone unit 2 as described above, the mixing of the target sound into the noise reference signal can be suppressed, but the mixing of the target sound into the noise reference signal cannot be completely eliminated. . Because, in an actual use environment, a reflected wave of a target sound is detected by the second microphone unit 2 due to an acoustic influence of a housing in which the microphone unit is mounted, a reflector around the microphone device, or the like. Because. Further, in addition to the influence of the reflected wave of the target sound, even if the blind spot of the directivity of the second microphone unit 2 is directed in the front direction, the direct wave of the target sound is detected to a small extent. Because there is). For the above reasons, the target sound component is mixed in the signal m2. Therefore, in the first embodiment, a canceller is configured by the determination unit 10, the adaptive filter unit 20, and the first signal subtraction unit 30. With this canceller, the component of the target sound mixed into the noise reference signal is removed. This makes it possible to obtain an ideal noise reference signal, that is, a noise reference signal in which the target sound is not mixed.

  In FIG. 1, the signal output from the first microphone unit 1 has a higher ratio of the target sound component than the noise component. The microphone device according to the first embodiment adaptively equalizes the signal m1 to the signal of the target sound component mixed in the signal m2 in the waveform equalization processing in the canceller. That is, the main signal is equalized to the signal of the target sound component included in the noise reference signal. This allows the canceller to operate with high accuracy.

  Furthermore, in the first embodiment, the adaptive filter unit 20 of the canceller performs the adaptive filter learning operation only when the target sound is generated sufficiently large. Specifically, the determination unit 10 detects whether the target sound is louder than the noise. The adaptive filter unit 20 performs an adaptive filter learning operation according to the detection result of the determination unit 10. This allows the filter coefficients of the adaptive filter unit 20 to stably converge. Note that the determination unit 10 needs to detect both the arrival direction and the level of the sound. The detailed configuration of the determination unit 10 will be described later (see FIG. 2).

  Next, the details of the operation together with the detailed configuration of each component of the microphone device will be described. FIG. 2 is a diagram illustrating a configuration of the determination unit illustrated in FIG. 2, the determination unit 10 includes a first signal level calculation unit 11, a second signal level calculation unit 12, a signal division unit 13, and a target sound arrival determination unit 14.

  In FIG. 2, a first signal level calculator 11 receives a signal m1 as an input, calculates a short-time average of signal levels of the signal m1, and outputs a first signal level x1a. The second signal level calculator 12 receives the signal m2 as an input, calculates a short-term average of the signal levels of the signal m2, and outputs a second signal level x2a. The signal divider 13 calculates a signal ratio (level ratio) between the first signal level x1a and the second signal level x2a. Specifically, the signal divider 13 outputs the signal ratio Va by dividing Va = x1a / x2a. Based on the output from the signal divider 13, the target sound arrival determiner 14 determines whether the target sound is generated sufficiently louder, that is, whether the target sound is louder than the noise. Specifically, the target sound arrival determination unit 14 compares the magnitude relationship between the signal ratio Va and the predetermined threshold th1, and outputs a determination result Vx indicating the magnitude relationship. More specifically, Vx is a value indicating that the signal ratio Va is greater than a predetermined threshold th1 (here, “1”), and Vx is a value indicating that the signal ratio Va is equal to or less than the predetermined threshold th1. It takes a binary value of a value indicating that there is (here, “0”).

  In FIG. 2, first, consider a case where the sound coming from the θ0 direction (front direction) is dominant. Here, “the sound coming from the θ0 direction is dominant” means that the sound coming from the θ0 direction is much larger than the sound coming from the other direction, and the sound coming from the other direction can be ignored. Means small. In this case, the θ0 direction coinciding with the front direction is the direction of the maximum sensitivity of the first microphone unit 1 and the direction of the minimum sensitivity of the second microphone unit. Therefore, the value of the first signal level x1a is large (relative to the case described later), and the value of the second signal level x2a is small (relative to the case described later). Therefore, in this case, the signal ratio Va (= x1a / x2a) has a large value (relative to the case described later).

  Next, consider the case where the sound coming from the θ1 direction is dominant. Here, the directivity of the first microphone 1 is unidirectional with the main directivity axis directed in the θ0 direction. The directional characteristics of the second microphone 2 are bidirectional with the main axis of directivity directed in the θ2 direction. Therefore, when the sound arriving from the θ1 direction is dominant, the value of the first signal level x1a decreases, and the value of the second signal level x2a becomes lower than when the sound arriving from the θ0 direction is dominant. The value increases. As a result, the signal ratio Va becomes smaller than when the sound coming from the θ0 direction is dominant. When the direction of the dominant sound shifts from the θ1 direction to the θ2 direction, the value of the first signal level x1a further decreases, and the value of the second signal level x2a further increases. As a result, the signal ratio Va becomes smaller than when the sound coming from the θ0 direction is dominant.

  Next, consider the case where the sound coming from the θ3 direction is dominant. Here, for both microphone units 1 and 2, the θ3 direction is a direction in which the directivity becomes a blind spot. Both the first signal level x1a and the second signal level x2a decrease, and as a result, the signal ratio Va does not become a large value.

  FIG. 3 is a diagram illustrating an example of a sound detection state when the dominant sound directions are the θ1 to θ3 directions. The waveforms of the first signal level x1a, the second signal level x2a, and the signal ratio Va are the signal waveforms shown in FIG. Here, by setting the threshold value th1 to the level shown in FIG. 3, it can be detected that the sound in the θ0 direction is dominant as the determination result Vx. That is, when the threshold value th1 is set to the level shown in FIG. 3, the value of the determination result Vx becomes “1” only when the sound in the θ0 direction is dominant. In the first embodiment, since the sound arriving from the front direction (θ0 direction) is set as the target sound, it can be detected from the value of Vx that the target sound is dominant. When not only the sound arriving from the θ0 direction but also the sound arriving from the θ1 direction is the target sound, the threshold may be set to th2 shown in FIG. Assuming that the threshold value is th2, the value of Vx becomes “1” not only when the sound in the θ0 direction but also in the θ1 direction is dominant.

  Next, the operation of removing the target sound mixed in the noise reference signal (signal m2) in the adaptive filter unit 20 and the signal subtraction unit 30 will be described. The adaptive filter unit 20 equalizes the signal m1 to the signal of the target sound component included in the signal m2 by the adaptive filter. That is, the adaptive filter unit 20 generates a signal of the target sound component included in the signal m2 from the signal m1. In addition, as a method of the adaptive filter, for example, an LMS method (learning identification method) or the like can be used. The signal subtraction unit 30 subtracts the signal generated by the adaptive filter unit 20 from the signal m2. As a result, the signal m3 becomes a noise reference signal from which the target sound component has been removed.

  Here, the adaptive filter unit 20 determines whether to learn the filter coefficient according to the determination result Vx by the determination unit 10. Specifically, when the determination unit 10 determines that the target sound is dominant, that is, when the determination result Vx indicates “1”, the adaptive filter unit 20 performs learning. On the other hand, when the determination unit 10 determines that the target sound is not dominant, that is, when the determination result Vx indicates “0”, the adaptive filter unit 20 does not perform learning.

  First, consider the case where the target sound is dominant. In this case, the adaptive filter unit 20 performs learning. Here, when the target sound is dominant, the noise can be neglected, so the second microphone unit 2 does not detect the noise, and the component of the target sound (a reflected wave of the target sound or a direct sound of the target sound). It can be considered that only the component (e.g., remaining wave cancellation) is detected. That is, the signal m2 can be regarded as including only the target sound component without including the noise component. In this case, adaptive filter section 20 may output signal m2 as a result of filtering signal m1. That is, the learning of the filter coefficient may be performed so that the signal m3 becomes 0. As a result of this learning, the adaptive filter unit 20 can obtain a filter coefficient for generating a signal of the target sound component included in the signal m2 based on the signal m1 with high accuracy.

  On the other hand, consider a case where the target sound is not dominant. In this case, the signal m2 includes a noise component having a magnitude that cannot be ignored in addition to the component of the target sound. Therefore, in this case, even if the adaptive filter unit 20 learns the filter coefficient so that the signal m3 becomes 0, it cannot obtain an appropriate filter coefficient. That is, it is impossible to obtain a filter coefficient for generating a signal of the target sound component included in the signal m2 based on the signal m1. Further, if learning is performed in such a case, the filter coefficients may diverge. For the above reason, the adaptive filter unit 20 should not learn the filter coefficients. Therefore, the adaptive filter unit 20 does not perform learning when the target sound is not dominant.

  As described above, learning of the adaptive filter is performed only when the magnitude of the target sound is larger than the surrounding noise by using the determination result of the determination unit 10. Thereby, the adaptive filter unit 20 can stably converge the filter coefficients.

  As described above, the microphone device according to the first embodiment first separates a target sound and a noise to some extent as preprocessing using the directional characteristics of each of the microphone units 1 and 2. Then, by using the canceller, the target sound component mixed in the noise reference signal, which cannot be completely suppressed by the configuration using the microphone units 1 and 2, is removed. As described above, the microphone device according to Embodiment 1 can obtain an ideal noise reference signal.

  Note that if a noise reference signal is to be obtained only by the configuration of the canceller without performing preprocessing using the directional characteristics of the microphone units 1 and 2, there are the following disadvantages. In an environment where noise is generated, it is difficult to detect a target sound, and therefore, there is a disadvantage that learning control accuracy is deteriorated. Further, since the target sound is not emphasized using the directivity of the microphone unit, there is a disadvantage that the correlation of the learning signal (target sound) is reduced and the convergence of the filter coefficient becomes difficult.

  Next, an operation of suppressing a noise component from a main signal (signal m1) by the noise suppression filter coefficient calculation unit 40 and the time-varying coefficient filter unit 50 will be described. Note that the same noise suppression effect as that of the noise suppression filter coefficient calculation unit 40 and the time-varying coefficient filter unit 50 can also be obtained by the configuration that performs the two-input type spectral subtraction method. However, when the spectrum subtraction method is performed, a frame processing for finally returning a spectrum to a waveform signal is required, so that a processing delay occurs. As a method for reducing the signal delay in the frame processing, it is conceivable to shorten the frame length or increase the frame overlap. However, the former is not realistic in that the frequency resolution is reduced, and the latter is not realistic in that the processing amount is increased. Therefore, the first embodiment employs a configuration using a time-varying coefficient filter, which is a method with a small processing delay.

  FIG. 4 is a diagram illustrating a configuration example of the noise suppression filter coefficient calculation unit 40. 4, the noise suppression filter coefficient calculation unit 40 includes a first frequency analysis unit 41, a second frequency analysis unit 42, a spectrum ratio calculation unit 43, a signal averaging unit 44, a signal multiplication unit 45, A filter transfer characteristic estimating unit 46 and an impulse response designing unit 47 are provided.

  In FIG. 4, the first frequency analysis unit 41 calculates a power spectrum X (ω) of the signal m1, which is the main signal. The second frequency analysis unit 42 calculates a power spectrum N1 (ω) of the signal m3 which is a noise reference signal. Here, each of the frequency analysis units 41 and 42 can be realized by using a known method that can derive the power of the frequency component, such as FFT, filter bank, wavelet transform, and DCT.

  The spectrum ratio calculation unit 43 receives the power spectrum X (ω) calculated by the first frequency analysis unit 41 and the power spectrum N1 (ω) calculated by the second frequency analysis unit 42, and H (ω) = X (ω) / N1 (ω) is derived. The signal averaging unit 44 receives as input the spectrum ratio H (ω) derived by the spectrum ratio calculation unit 43 and the determination result Vx by the determination unit 10. Then, the time average Ha (ω) is calculated for each frequency component when the ambient noise is more dominant than the target sound (that is, when the value of Vx is “0”). The signal multiplying unit 45 multiplies the power spectrum N1 (ω) calculated by the second frequency analyzing unit 42 and the time average Ha (ω) calculated by the signal averaging unit 44 for each frequency component. Then, the multiplication result is output as Nx (ω). Note that, due to the difference in the directivity pattern, the characteristics of the microphone unit, and the like, the shape and level of the spectrum of the noise component other than the target sound component included in the spectrum X (ω) of the main signal are different from the spectrum N1 of the noise reference signal. It does not always equal the shape and level of (ω). The spectrum ratio calculating section 43, the signal averaging section 44, and the signal multiplying section 45 described above include the spectrum of the noise component other than the target sound component included in the spectrum X (ω) of the main signal and the spectrum N1 of the noise reference signal. (Ω). Therefore, Nx (ω) obtained as a result of the multiplication by the signal multiplying unit 45 becomes a noise component included in the spectrum X (ω) of the main signal. Therefore, this Nx (ω) is called an estimated noise spectrum Nx (ω).

  The filter transfer characteristic estimation unit 46 receives the power spectrum X (ω) calculated by the first frequency analysis unit 41 and the estimated noise spectrum Nx (ω) calculated by the signal multiplication unit 45, and Is calculated. The transfer characteristic Hw (ω) of the noise suppression filter can be obtained by Hw (ω) = (X (ω) −Nx (ω)) / X (ω) based on, for example, the Wiener filter method.

  The impulse response designing unit 47 sets the transfer characteristic Hw (ω) calculated by the filter transfer characteristic estimating unit 46 as a target characteristic, and outputs a filter coefficient hw (n) so as to approach the target characteristic every sample. .

  The time-varying coefficient filter unit 50 filters the signal m1 according to the filter coefficient hw (n) output from the impulse response design unit 47, and generates an output signal y of the microphone device. Hereinafter, a specific configuration example of the time-varying coefficient filter unit 50 will be described with reference to FIGS.

  FIG. 5 is a diagram illustrating a configuration example of the time-varying coefficient filter unit 50. In FIG. 5, the time-varying coefficient filter unit 50 includes n signal delay units, n + 1 signal amplification units, and n signal addition units. In FIG. 5, the first signal delay unit 501, the second signal delay unit 502, the n-th signal delay unit 503, the first signal amplifying unit 504, the second signal amplifying unit 505, and the n-th signal delay unit Only the signal amplification unit 506, the first signal addition unit 508, and the n-th signal addition unit 509 are shown.

  In FIG. 5, each signal delay unit is cascaded and delays an input signal by one sample. Each signal amplifying unit amplifies and outputs an input signal. The first signal amplifying unit 504 amplifies the signal m1 input to the time-varying coefficient filter unit 50. Second signal amplifying section 505 amplifies the signal output from first signal delay section 501. Thereafter, the signal amplifying unit downstream of the second signal amplifying unit 505 performs the same operation as the second signal amplifying unit 505. That is, the (i + 1) th signal amplifier amplifies the signal output from the i-th signal delay unit (i is an integer from 1 to n). First signal addition section 508 adds the signal output from first signal amplification section 504 and the signal output from second signal amplification section 505. The second signal addition unit (not shown) adds the signal output from the first signal addition unit 508 and the signal output from the third signal amplification unit (not shown). After that, the signal addition unit subsequent to the second signal addition unit (not shown) performs the same operation as the second signal addition unit. That is, the j-th signal addition unit adds the signal output from the j-1st signal addition unit and the signal output from the (i + 1) -th signal amplification unit (j is an integer from 2 to n) ). Then, the signal output from the n-th signal addition unit 509 becomes the output signal y. The configuration shown in FIG. 5 is a configuration of a general FIR filter, and each coefficient of the first to (n + 1) th signal amplifiers changes according to the filter coefficient hw (n) from the impulse response design unit 47. .

  FIG. 6 is a diagram illustrating another configuration example of the time-varying coefficient filter unit 50. 6, the time-varying coefficient filter unit 50 includes n band-pass filters, n signal amplifying units, and a signal adding unit 517. In FIG. 6, the first band-pass filter 511, the second band-pass filter 512, the n-th band-pass filter 513, the first signal amplifying unit 514, the second signal amplifying unit 515, and the n-th Only the signal amplifier 516 and the signal adder 517 are shown.

  In FIG. 6, each band-pass filter is provided in parallel at the subsequent stage of the input signal, and divides the band of the signal m1 input to the time-varying coefficient filter unit 50 into n bands and outputs the band. Each signal amplifier amplifies a signal output from each bandpass filter. The signal adder 517 adds the signals output from the respective signal amplifiers, and outputs the added result as an output signal y. The amplification factor of each signal amplifying unit can be determined based on the transfer function Hw (ω) output from the filter transfer characteristic estimating unit 46. With the above configuration, the same effect as in FIG. 5 can be obtained.

  FIG. 7 is a diagram showing a specific example of each signal shown in FIG. Specifically, a signal m1 output from the first microphone unit 1, a signal m2 output from the second microphone unit 2, a signal m3 output from the first signal subtraction unit 30, and a time-varying coefficient filter A specific example of the output signal y output from the unit 50 will be described. As shown in FIG. 7, the signal m3 is a signal including only components other than the target sound, that is, only noise components, from which the influence of the reflected sound and the like is removed from the signal m2. Further, by performing a filtering process using the main signal m1 and the noise reference signal m3 in the time-varying coefficient filter section 50, only the target sound can be extracted as the output signal y. As is clear from comparison of the output m1 of the conventional directional microphone unit with the output signal y of the microphone device of the first embodiment, according to the microphone device of the first embodiment, the target sound is generated. Regardless of whether the target sound is present or the target sound is not generated, the ambient noise can be significantly suppressed.

  Depending on the positional relationship between the first microphone unit 1 and the second microphone unit 2 and the circuit of each component provided after each of the microphone units 1 and 2, it may be necessary to satisfy the causal rule for the convergence of the adaptive filter. For the purpose, a signal delay unit may be provided between the signal subtraction unit 30 and the second microphone unit 2. The amount of delay in the signal delay unit is determined with reference to a value obtained by dividing the distance between the microphone units 1 and 2 by the speed of sound or more.

  Further, in the first embodiment, a unidirectional microphone unit is used as the first microphone unit 1, but an omnidirectional microphone or a superdirectional microphone may be used.

  In the above description, the determination unit 10 outputs a numerical value expressed in binary as the determination result Vx. Here, the determination unit 10 may output the signal ratio Va expressed in multiple values. Further, in this case, the adaptive filter unit 20 changes the learning speed according to the determination result (signal ratio Va). Specifically, when the signal ratio Va is larger than the threshold value, the adaptive filter unit 20 increases the learning speed as the signal ratio Va increases. More specifically, the adaptive filter unit 20 approaches the value of the step gain parameter to 0.5 as the signal ratio Va increases. On the other hand, when the signal ratio Va is equal to or smaller than the threshold, the adaptive filter unit 20 does not perform learning. More specifically, the value of the step gain parameter is set to 0.

  As described above, the microphone device according to Embodiment 1 can obtain an ideal noise reference signal even in a noise environment and in a reflected sound field. Therefore, the noise suppression unit using the main signal and the noise reference signal can greatly improve the sound collection S / N as compared with the conventional directional microphone. Further, the microphone device according to the first embodiment can reduce the processing delay by employing a method using a time-varying coefficient filter as the noise suppression method, as compared with the case of using the spectrum subtraction method. Therefore, the microphone device according to the first embodiment can be applied to applications requiring low delay processing, such as loudspeaker applications and speech applications.

(Embodiment 2)
Next, a microphone device according to the second embodiment will be described with reference to FIGS. Note that the microphone device according to the first embodiment aims at suppressing noise mixed in when a target sound is detected. The microphone device according to the second embodiment aims to correct frequency characteristic distortion of a target sound due to detection of a reflected wave of the target sound.

  8, the microphone device includes a first microphone unit 1, a second microphone unit 2, a determination unit 10, an adaptive filter unit 20, a signal subtraction unit 30, a reflection information calculation unit 60, a reflection correction unit And a unit 70. In FIG. 8, the same components as those in the first embodiment are denoted by the same reference numerals as those in FIG. 1, and detailed description is omitted.

  In FIG. 8, a filter coefficient of the adaptive filter unit 20 is input to the reflection information calculation unit 60. The reflection information calculation unit 60 estimates the presence or absence of a reflection object, the distance, and the degree of influence using the input filter coefficient. The reflection correction unit 70 receives the signal m1 as input, and corrects the frequency characteristic distortion generated in the signal m1 due to the influence of the reflection of the target sound based on the estimation result of the reflection information calculation unit 60.

  Hereinafter, the operation of the microphone device according to Embodiment 2 will be described.

  In the microphone device shown in FIG. 8, the signal m1 is the main signal. Here, when the directivity of the first microphone unit 1 is unidirectional, the directivity of the first microphone unit 1 is not sharp enough to remove the reflected wave of the target sound. Therefore, when a reflected object is present near the microphone device, the reflected wave is simultaneously picked up in addition to the direct wave of the target sound, so that the frequency characteristic of the detected sound is the interference between the direct wave of the target sound and the reflected wave. Will be disturbed. The microphone device according to the second embodiment corrects the frequency characteristic distorted by the influence of the reflection of the target sound by using the fact that the information of the reflected wave appears in the filter coefficient of the adaptive filter unit 20. This enables automatic correction of the frequency characteristics of the detected sound.

  As described above, the adaptive filter unit 20 generates a signal of a target sound component mixed in the signal m2, that is, a signal of a residual component of the target sound due to imperfect directivity and a reflected wave component of the target sound. That is, the transfer characteristic (impulse response) from the signal m1 containing many components of the direct wave of the target sound to the signal m2 containing many components of the reflected wave of the target sound is expressed by the filter coefficient of the adaptive filter unit 20. Will be. Therefore, by detecting the peak of the coefficient from the filter coefficient, the time difference dt (sec) between the time at which the direct wave of the target sound arrives at the position of the microphone unit and the time at which the reflected wave arrives, and the reflected wave are represented. The peak level Lr and the intensity of reflection can be understood. Further, from the time difference dt, the distance difference dt × c (where c is the sound speed) between the path where the reflected wave of the target sound arrives and the path where the direct wave arrives can be found.

  Here, for a sound having a frequency whose wavelength is equal to the distance difference (the wavelength λ satisfies the relationship of λ = dt × c), the direct wave and the reflected wave are added in the same phase, and are detected by the microphone unit. The sound pressure level is increased. Conversely, for a sound having a frequency whose wavelength is equal to one half of the distance difference (the wavelength λ satisfies the relationship of λ / 2 = dt × c), the direct wave and the reflected wave have opposite phases. The sound pressure level detected by the microphone unit decreases, and a dip occurs in the frequency characteristics of the main signal. Assuming that perfect reflection occurs on the reflecting surface, the signal output from the first microphone unit 1 emphasizes a harmonic portion having a fundamental frequency of fa (= c / λ = 1 / dt). A comb-like frequency characteristic appears.

  FIG. 9 is a diagram illustrating a difference in the internal state of the microphone device between when there is a reflective object and when there is no reflective object. In FIG. 9, the positional relationship between the microphone unit, the target sound source (speaker), and the reflective object, and the value of the adaptive filter coefficient hadf (n) in the adaptive filter unit 20 for the case where there is a reflective object and the case where there is no reflective object And the frequency characteristics of the signal m1.

  In FIG. 9, when there is no reflector near the speaker and the microphone unit as shown in (a1), as shown in (a2), the filter coefficient of the adaptive filter unit 20 is not affected by the reflected wave. It does not appear. Further, as shown in (a3), the shape of the frequency characteristic of the main signal becomes relatively flat. On the other hand, in a state where there is a reflector near the speaker and the microphone as shown in (b1), as shown in (b2), the filter coefficient of the adaptive filter unit 20 has a large value in the time difference dt portion. Become. Further, as shown in (b3), the frequency characteristic of the main signal is also distorted according to the positional relationship between the microphone, the target sound source, and the reflector.

As described above, the time difference dt and the degree of influence Lr can be calculated from the coefficient peak of the adaptive filter. Further, using these, it is possible to estimate the correction amount of the frequency characteristic that is distorted by the influence of the reflected wave. In practice, especially in a high-frequency range, it cannot be considered that complete reflection occurs on the reflection surface. If it cannot be considered that complete reflection occurs on the reflecting surface, it is conceivable to design a deconvolution filter on the assumption of the reflecting characteristics of the reflecting surface. In addition, focusing only on the low-frequency characteristic for simplicity, a frequency (fa = 1 / dt) in which one wavelength is equal to the distance difference, a frequency (fb = 1 / 2dt) in which 1/2 wavelength is equal to the distance difference, and the like. For the frequency, for example, a correction gain is calculated by the following equation.
Center frequency fa: correction gain = −β1 · 20log (1 + α1 · Lr) (dB)
Center frequency fb: correction gain = + β2 · 20log (1−α2 · Lr) (dB)
In this case, the correction characteristic Hr (ω) of the reflection correction unit 70 can be realized by an equalizer that can adjust the center frequency, the bandwidth, and the gain based on the information from the reflection information calculation unit 60.

  When the use environment can be limited, for example, when the microphone device is used for voice recognition of car navigation, the detection accuracy of the filter coefficient of the adaptive filter unit 20 can be improved. Specifically, only the initial reflection component is targeted, and the search range of the maximum value of the filter coefficient is limited based on the reflected wave delay calculated from the reflection surface position.

  Depending on the directivity type of the microphone unit, the maximum value of the filter coefficient may depend on the direction of the reflected wave, depending on the polarity of the directional lobe, whether the peak due to the reflected wave is positive or negative. In such a configuration, it is necessary to search for the maximum value for the absolute value of the coefficient.

  As described above, according to the second embodiment, it is possible to correct the frequency characteristic that is distorted by the influence of the reflected wave of the target sound. Therefore, it is possible to realize a microphone device that can stably obtain a flat sound pressure sensitivity versus frequency characteristic in any use environment (sound field). Therefore, according to the second embodiment, it is possible to improve the sound quality in a telephone call or loudspeaking. In addition, particularly in speech recognition applications, the frequency characteristic distortion caused by the reflected wave is one of the factors of erroneous recognition. However, according to the configuration of the second embodiment, a stable high speech is obtained regardless of the presence or absence of a nearby reflective object. The recognition rate can be realized.

(Embodiment 3)
Next, a microphone device according to the third embodiment will be described with reference to FIGS. The microphone device according to the third embodiment has a configuration in which the configuration of the first embodiment and the configuration of the second embodiment are combined.

  FIG. 10 is a block diagram showing a configuration of the microphone device according to Embodiment 3. In FIG. 10, the microphone device includes a first microphone unit 1, a second microphone unit 2, a determination unit 10, an adaptive filter unit 20, a signal subtraction unit 30, a noise suppression filter coefficient calculation unit 40, It includes a time-varying coefficient filter section 50, a reflection information calculation section 60, and a reflection correction section 70. In FIG. 10, the same components as those in Embodiment 1 or 2 are denoted by the same reference numerals as those in FIG. 1 or FIG. 8, and detailed description is omitted.

  The difference between the configuration shown in FIG. 10 and the configuration shown in FIG. 8 is that the noise suppression filter coefficient calculation unit 40 and the time-varying coefficient filter unit 50 shown in FIG. 1 are provided after the configuration shown in FIG. . Thus, the microphone device shown in FIG. 10 can correct distortion of frequency characteristics due to a reflected wave and perform noise suppression.

  FIG. 11 is a block diagram showing another configuration of the microphone device according to the third embodiment. In FIG. 11, the microphone device includes a first microphone unit 1, a second microphone unit 2, a determination unit 10, an adaptive filter unit 20, a signal subtraction unit 30, a time-varying coefficient filter unit 50, a reflection unit An information calculation unit 60, a reflection correction unit 70, and a noise suppression and reflection inverse characteristic filter coefficient estimation unit 80 are provided. The configuration shown in FIG. 11 is a configuration in which the processing amount is reduced by superimposing the characteristics of the reflection correction unit 70 on the characteristics of the time-varying coefficient filter unit 50.

  The operation of the configuration shown in FIG. 11 is different from the operation of the configuration shown in FIG. The noise suppression and reflection inverse characteristic filter coefficient estimation unit 80 receives the signal m1 (main signal), the signal m3 (noise reference signal), and the signal output from the reflection information calculation unit 60 as inputs. Then, based on these signals, a noise suppression filter characteristic Hw (ω) = (X (ω) −Nx (ω)) / X (ω) and a reflection inverse characteristic Hr (ω) are calculated. Further, a filter coefficient having {Hw (ω) · Hr (ω)} as a target characteristic is output to the time-varying coefficient filter unit 50. This makes it possible to simultaneously perform the processing of correcting the distortion of the frequency characteristic due to the reflected wave and the processing of suppressing the noise.

  As described above, according to Embodiment 3, it is possible to obtain an ideal noise reference signal from which the target sound has been removed, as in Embodiment 1. Further, as in the second embodiment, two-input noise suppression processing using a main signal and a noise reference signal and correction processing of frequency characteristic distortion due to the influence of a reflected wave can be performed simultaneously. As a result, whether the surrounding environment is a noisy environment or a reflected sound field, a high S / N and flat frequency characteristic can be obtained, and the effect of improving the voice quality of a telephone call or loudspeaking, The effect of improving the recognition rate of voice recognition is obtained.

(Embodiment 4)
Next, a microphone device according to Embodiment 4 will be described with reference to FIGS. In the fourth embodiment, of the sounds arriving at the microphone device in all directions, the direction regarded as the target sound is changed.

  FIG. 12 is a block diagram showing a configuration of the microphone device according to Embodiment 4. 12, the microphone device further includes a detection threshold setting unit 90 in addition to the configuration shown in FIG. In FIG. 12, the same components as those in the third embodiment are denoted by the same reference numerals as those in FIG. 11, and detailed description is omitted.

  The detection threshold setting unit 90 sets a threshold value used in the determination unit 10. That is, the configuration of the fourth embodiment differs from the configuration of the third embodiment in that the threshold value set in the determination unit 10 can be controlled.

  In FIG. 12, the threshold value set in the determination unit 10 can be changed. By changing the threshold value, it is possible to change the angle at which the direction in which the sound regarded as the target sound arrives from the front direction on both the left and right sides. That is, by changing the threshold value, it is possible to control the range of angles at which sound can be collected as the target sound.

  For example, consider the case where the threshold is set to th1 by the detection threshold setting unit 90 (see FIG. 3). In this case, the sound coming from the θ1 direction (see FIGS. 2 and 3) is not regarded as the target sound. That is, the sound arriving from the θ1 direction is regarded as noise, and the signal m3, which is the noise reference signal, includes the sound component arriving from the θ1 direction. As a result, in the final output, the sound arriving from the θ1 direction is suppressed.

  On the other hand, when the threshold is set to th2 (see FIG. 3), the sound coming from the θ1 direction is regarded as the target sound. In this case, the signal m3, which is the noise reference signal, does not include a sound component arriving from the θ1 direction. As a result, in the final output, the sound arriving from the θ1 direction is output as the target sound.

  As described above, by controlling the threshold value of the determination unit 10, it is possible to control the angle range in which the microphone device can collect sound. However, the angle range is limited to a certain angle range with respect to the directional blind spot direction of the second microphone unit 2, that is, the front direction.

  FIG. 13 is a diagram illustrating a directivity pattern of the microphone device. FIG. 13A is a diagram illustrating a directivity pattern of the signal m1. FIG. 13B is a diagram illustrating a directivity pattern of the output signal y of the microphone device when the threshold is set to th2. FIG. 13C is a diagram illustrating a directivity pattern of the output signal y of the microphone device when the threshold is set to th1. In FIG. 13B, the angle range in which the microphone device can collect sound is wider than in FIG. 13C. For example, when the threshold value is set to th2, the sound coming from the angle θ1 is determined as the target sound. Further, the sensitivity is greatly attenuated in a portion outside the range. On the other hand, in FIG. 13C, the angle range in which the microphone device can collect sound is narrow, and a very sharp directional characteristic is realized. In this case, the sound coming from the angle θ1 is not determined as the target sound.

  As described above, according to the fourth embodiment, the sharpness of the directivity of the microphone device can be changed by changing the threshold value of determination unit 10. In general, the directivity of a microphone is more difficult to form a sharp main beam than to form a sharp blind spot. However, according to the fourth embodiment, a microphone device having an unprecedented sharp directivity is realized. can do.

  Here, in actual use, sharpness of directivity and ease of use of the microphone device are opposite to each other. When using a microphone device having a sharp directivity, the user must use the microphone device with a strong awareness of the front direction. Therefore, in order to achieve both ease of use and noise suppression performance, the microphone device has a constant sensitivity characteristic from the front to a certain angle range, and has a directional characteristic such that sensitivity attenuation in other directions is large. It is desirable. Further, it is desirable that the angle range in which sound can be collected can be set freely according to the use of the microphone device and the sound collecting situation. According to the fourth embodiment, the directivity of the microphone device changes as shown in FIG. As is clear from FIG. 13, the microphone device according to the fourth embodiment can achieve both ease of use as a microphone device and high noise removal capability.

(Embodiment 5)
Next, a microphone device according to Embodiment 5 will be described with reference to FIG. Note that the microphone devices according to Embodiments 1 to 4 have a unidirectional microphone unit and a bidirectional microphone unit arranged close to each other, and a signal output from each microphone unit is defined as a main signal and a noise reference signal. Configuration. The advantages of this configuration are that it can be reduced in size and can be realized at low cost because processing such as directivity synthesis is not required.

  On the other hand, a video movie and other devices having a sound collection function often use a plurality of microphone units having omnidirectionality or directivity of the same characteristic due to mounting problems and performance problems, and these microphone units are used. In some cases, directivity may be formed by combining signals output from the devices. In the process of performing directivity synthesis on signals from a plurality of microphone units, a certain interval (usually 1 cm to 5 cm) is required as an interval between the microphone units due to a problem such as circuit noise. Therefore, the method for performing the processing is disadvantageous in terms of miniaturization compared to the first to fourth embodiments. However, the method of performing the processing has advantages in terms of mounting, such as a high degree of design freedom of directivity and a point that variable characteristics using digital processing can be used.

  Therefore, in the fifth embodiment, using a plurality of microphone units (two in the fifth embodiment) having the same directional characteristics and the directivity combining unit 100, a main signal corresponding to the signal m1 is obtained. A configuration for obtaining a noise reference signal corresponding to the signal m2 is adopted.

  FIG. 14 is a diagram illustrating a part of the configuration of the microphone device according to the fifth embodiment. In FIG. 14, the microphone device includes a third microphone unit 3, a fourth microphone unit 4, and a directivity synthesis unit 100. Note that any of the configurations of Embodiments 1 to 4 is used for the configuration after obtaining the signal m1 and the signal m2.

  In FIG. 14, each of the microphone units 3 and 4 is arranged on an axis (a dashed line shown in FIG. 14) facing the front direction. The distance between each microphone unit 3 and 4 is d. Each of the microphone units 3 and 4 is arranged so that its directivity main axis faces the front direction.

  Further, the directivity combining unit 100 includes a first signal delay unit 101, a first signal subtraction unit 103, a second signal delay unit 102, and a second signal subtraction unit 104. First signal delay section 101 delays a signal output from fourth microphone unit 4. The second signal delay unit 102 delays a signal output from the third microphone unit 3. The first signal subtraction unit 103 subtracts the signal output from the first signal delay unit 101 from the signal output from the third microphone unit 3. As a result, a signal m1 is obtained. The second signal subtraction unit 104 subtracts the signal output from the second signal delay unit 102 from the signal output from the fourth microphone unit 4. As a result, a signal m2 is obtained.

  Further, by setting the delay amount τ1 of the first signal delay unit 101 to 0 ≦ τ1 ≦ d / c (where c is the sound speed), the secondary sound pressure gradient type super-directivity in which the directivity main axis is in the front direction. The sex characteristic can be obtained as the signal m1. Also, by setting the signal delay amount τ2 of the second signal delay unit 102 to τ2 = d / c, a signal that forms a directional blind spot in the front direction (a microphone that forms a directional blind spot in the front direction) The resulting signal from the unit) m2 can be obtained.

  With the above configuration, by realizing the super directivity in the characteristic of the signal m1 in advance, it is possible to realize sharp directivity and noise suppression performance that greatly exceeds the conventional super directivity microphone by combining with the subsequent noise suppression processing. it can.

(Embodiment 6)
Next, a microphone device according to Embodiment 6 will be described with reference to FIG. Embodiment 6 employs a configuration in which a main signal and a noise reference signal are obtained using a plurality of microphone units having the same directional characteristics, as in Embodiment 5.

  FIG. 15 is a diagram illustrating a part of the configuration of the microphone device according to the sixth embodiment. The microphone device includes a third microphone unit 3, a fourth microphone unit 4, and a directivity synthesis unit 100. Each of the microphone units 3 and 4 is arranged on an axis (a dashed line shown in FIG. 15) perpendicular to a straight line facing the front direction (a dotted line shown in FIG. 15). Each of the microphone units 3 and 4 is arranged so that its directivity main axis faces the front direction. Note that any of the configurations of Embodiments 1 to 4 is used for the configuration after obtaining the signal m1 and the signal m2.

  In FIG. 15, the directivity synthesis unit 100 includes a first signal addition unit 105 and a second signal subtraction unit 104. The first signal addition unit 105 adds signals output from the microphone units 3 and 4. As a result, a signal m1, which is a main signal, can be obtained. The second signal subtraction unit 104 subtracts a signal output from the third microphone unit 3 from a signal output from the fourth microphone unit 4. Thereby, signal m2 which is a noise reference signal can be obtained.

  In FIG. 15, when the interval between the microphone units 3 and 4 is narrow to some extent, the directivity of the signal m1 is not much different from the case of the microphone unit alone (Embodiments 1 to 4) except for the high-frequency characteristics. Therefore, in the configuration shown in FIG. 15, it is not possible to obtain sharp directivity as compared with the configuration shown in FIG. 14, but on the other hand, an effect of reducing vibration noise and circuit noise is obtained. Further, since the sound arriving from the front direction is detected by the microphone units 3 and 4 in the same phase, it is possible to obtain a signal m2 in which a directional blind spot is formed in the front direction.

(Embodiment 7)
Next, a microphone device according to Embodiment 7 will be described with reference to FIG. Embodiment 7, as in Embodiment 5, employs a configuration in which a main signal and a noise reference signal are obtained using a plurality of microphone units having the same directional characteristics.

  FIG. 16A is a diagram illustrating a part of the configuration of the microphone device according to the seventh embodiment. The microphone device includes a third microphone unit 3, a fourth microphone unit 4, and a directivity synthesis unit 100. The arrangement of the microphone units 3 and 4 is the same as the arrangement shown in FIG. Note that any of the configurations of Embodiments 1 to 4 is used for the configuration after obtaining the signal m1 and the signal m2.

  In FIG. 16A, the directivity synthesis unit 100 includes a signal delay unit 111, a second signal subtraction unit 104, a signal amplification unit 150, and a first signal subtraction unit 103. The signal delay unit 111 receives the signal output from the third microphone unit 3 and delays the signal. Second signal subtraction section 104 subtracts the signal output from signal delay section 111 from the signal output from fourth microphone unit 4. As a result, a signal m2 that is a noise reference signal can be obtained. The signal amplification unit 150 multiplies the signal output from the signal delay unit 111 by a constant. The first signal subtraction unit 103 subtracts a signal output from the signal amplification unit 150 from a signal output from the fourth microphone unit 4. As a result, the signal m1, which is the main signal, can be obtained.

  16A, the difference between the process of obtaining the signal m1 and the process of obtaining the signal m2 is that the signal amplifying unit 150 exists in the process of obtaining the signal m1. The blind spot direction of the directivity of the signal m1 and the signal m2 is determined by the delay amount τ1 of the signal delay unit 111. For example, when τ1 = 0, the directional blind spot is in the front direction, and when τ1 = d / c, the directional blind spot is in a direction perpendicular to the front direction. Here, the delay amount τ1 is set so that a blind spot is formed in the direction of the target sound. As a result, the signal m1 and the signal m2 contain more components of the sound arriving from directions other than the target sound direction than components of the target sound.

  Here, it is preferable that the directivity pattern formed by the directivity synthesis unit 100 has a large sensitivity difference between the signal m1 and the signal m2 in the direction of the target sound. On the other hand, it is preferable that there is no difference in sensitivity characteristics between the signal m1 and the signal m2 in directions other than the direction of the target sound. This is because, in a situation where noise is coming from multiple directions at the same time, in order to suppress the noise component mixed into the main signal based on the noise reference signal, the output of the spectrum ratio calculation unit 43 shown in FIG. This is because it needs to be constant regardless of the direction from which the noise arrives. That is, if the output of the spectrum ratio calculator 43 changes depending on the direction of arrival of noise, only the estimated noise spectrum Nx (ω) in a specific direction can be accurately obtained. Therefore, it is preferable that the directivity patterns of the signal m1 and the signal m2 have different shapes only in the blind spot portion of the directivity, and have the same shape in other portions.

  Here, if the sensitivity balance between the third microphone unit 3 and the fourth microphone unit 4 is lost when the signals from the microphone units 3 and 4 are subtracted, the zero point that requires the most accuracy, that is, the directivity The sensitivity in the blind spot increases. By utilizing this property and providing the signal amplifier 150 on the signal m1 side and setting the signal amplification factor to about 0.85, a directivity pattern as shown in FIG. 16B can be obtained. FIG. 16B is a diagram illustrating the directivity pattern of the signal m1 and the signal m2 in FIG. As shown in FIG. 16B, in the seventh embodiment, it is possible to obtain a directivity pattern in which the shape is different only in the blind spot of directivity and the shape is almost the same in other portions.

  As described above, according to the seventh embodiment, it is possible to obtain signals m1 and m2 having different sensitivity characteristics only in the target sound direction. Therefore, a good suppression effect can be obtained in the noise suppression processing in the subsequent stage.

(Embodiment 8)
Next, a microphone device according to Embodiment 8 will be described with reference to FIG. Embodiment 8, as in Embodiment 5, employs a configuration in which a main signal and a noise reference signal are obtained using a plurality of microphone units having the same directional characteristics.

  FIG. 17A is a diagram illustrating a part of the configuration of the microphone device according to the eighth embodiment. In FIG. 17A, the directivity synthesis section 100 further includes an angle setting section 160 and a second signal delay section 112 in addition to the configuration shown in FIG. Note that any of the configurations of Embodiments 1 to 4 is used for the configuration after obtaining the signal m1 and the signal m2.

  The configuration illustrated in FIG. 17A is different from the configuration illustrated in FIG. 16A in that an angle setting unit 160 is further provided and a second signal delay unit 112 is provided at a stage subsequent to the fourth microphone unit 4. . Note that the basic operation in FIG. 17A is the same as that in FIG. 17A is different from the operation in FIG. 16A in that the angle setting unit 160 can change the target sound direction.

  The angle setting section 160 can change the signal delay amount τ1 of the first signal delay section 111 in a range of 0 ≦ τ1 ≦ 2d / c (where d is the interval between the microphone units and c is the sound speed). And Here, when the second signal delay unit 112 is not provided, even if the signal delay amount τ1 of the first signal delay unit 111 is changed in the above range, the signal delay amount τ1 is in a range from 0 ° to + 90 ° with respect to the front direction. Only the target sound direction can be changed. Therefore, the second signal delay unit 112 is provided, and by setting the signal delay amount τ2 to τ2 = d / c, the target sound direction is changed within a range of ± 90 ° with respect to the front direction.

  As described above, in the eighth embodiment, the sound pickup direction (target sound direction) of the microphone device can be made variable. For example, the directivity pattern shown in FIG. 17B can be realized, and the directivity pattern shown in FIG. 17C can be realized by changing the signal delay amount of the signal delay unit. It is possible. The variable delay characteristic can be easily realized by configuring the signal delay unit with an all-pass filter H (ω) = (A + z−1) / (1 + A · z−1) and setting the coefficient A to 0 ≦ A <1. can do. When changing the signal delay amount, the coefficient A is changed by the angle setting unit 160. When a large amount of delay or linearity of the delay frequency characteristic is required, a secondary all-pass filter and / or an all-pass filter may be connected in cascade.

(Embodiment 9)
Next, a microphone device according to Embodiment 9 will be described with reference to FIG. Embodiment 9, as in Embodiment 5, employs a configuration in which a main signal and a noise reference signal are obtained using a plurality of microphone units having the same directional characteristics.

  FIG. 18A is a diagram illustrating a part of the configuration of the microphone device according to the ninth embodiment. The microphone device includes a third microphone unit 3, a fourth microphone unit 4, a directivity synthesis unit 100, and an angle setting unit 160. The arrangement of the microphone units 3 and 4 is the same as the arrangement shown in FIG. Note that any of the configurations of Embodiments 1 to 4 is used for the configuration after obtaining the signal m1 and the signal m2.

  In FIG. 18A, the directivity synthesis unit 100 includes a third signal delay unit 121, a first signal delay unit 101, a fourth signal delay unit 122, a second signal delay unit 102, It includes a first signal subtraction unit 103 and a second signal subtraction unit 104. Third signal delay section 121 delays a signal output from third microphone unit 3. First signal delay section 101 delays a signal output from fourth microphone unit 4. First signal subtraction section 103 subtracts the signal output from first signal delay section 101 from the signal output from third signal delay section 121. As a result, the signal m1, which is the main signal, can be obtained. The fourth signal delay unit 122 delays a signal output from the fourth microphone unit 4. The second signal delay unit 102 delays a signal output from the third microphone unit 3. Second signal subtraction section 104 subtracts the signal output from second signal delay section 102 from the signal output from fourth signal delay section 122. Thereby, signal m2 which is a noise reference signal can be obtained. The angle setting unit 160 controls the signal delay amount of the first signal delay unit 101 and the signal delay amount of the second signal delay unit 102 independently.

  In FIG. 18A, the configuration on the signal m1 side is configured symmetrically with respect to the configuration on the signal m2 side. As a result, the directivity pattern of the signal m1 and the directivity pattern of the signal m2 are controlled independently, so that the directivity patterns of the signal m1 and the signal m2 can be designed so as to emphasize the sensitivity in the target sound direction. it can. Specifically, as shown in FIG. 18B, the directivity pattern of the signal m1 is set to have directivity as high as possible in the target sound direction and a noise suppression effect can be obtained. Further, as shown in FIG. 18C, the directivity pattern of the signal m2 is formed so that the blind spot direction of the directivity matches the target sound direction.

  As described above, in the ninth embodiment, the noise suppression processing of the subsequent stage is used as an auxiliary, and the noise is actively suppressed by the directivity synthesis of the preceding stage. Therefore, in the ninth embodiment, the directivity pattern of the signal m1 is preferentially formed. Here, since the directivity synthesis is a linear process, there is a feature that sound waveform distortion and the like hardly occur. On the other hand, the noise suppression processing is a non-linear processing in which the filter coefficient changes with time, so that there may be a case where voice waveform distortion is caused by errors of various estimation units such as a noise spectrum. From this point of view, whether to use the directivity patterns shown in FIGS. 17B and 17C or the directivity patterns shown in FIGS. 18B and 18C depends on the usage environment (purpose). It is preferable to appropriately select according to the loudness of the sound, the ambient noise level, reflection, reverberation, etc., the use (call, voice recognition, recording, etc.), the required amount of noise suppression, and the like.

(Embodiment 10)
Next, a microphone device according to Embodiment 10 will be described with reference to FIG. Embodiment 10 aims at obtaining a main signal and a noise reference signal required for the noise suppression processing of the present invention in a device in which the directivity main axes of two microphone units are provided in different directions.

  FIG. 19 is a diagram illustrating a part of the configuration of the microphone device according to the tenth embodiment. The microphone device includes a third microphone unit 3, a fourth microphone unit 4, and a directional resynthesis unit 200. Note that any of the configurations of Embodiments 1 to 4 is used for the configuration after obtaining the signal m1 and the signal m2.

  In FIG. 19, the positions where the microphone units 3 and 4 are arranged are the same as the positions shown in FIG. However, in FIG. 19, the directivity main axis of the third microphone unit 3 is directed in a direction rotated by a predetermined angle with respect to the front direction. The directivity main axis of the fourth microphone unit 4 is directed in a direction rotated by a predetermined angle with respect to the front direction (a rotation direction opposite to that of the third microphone unit 3). Here, the signal output from the third microphone unit 3 is called a right channel signal, and the signal output from the fourth microphone unit 4 is called a left channel signal.

  In FIG. 19, the directivity re-synthesis unit 200 includes a signal addition unit 205 and a signal subtraction unit 204. The signal adding unit 205 adds the right channel signal and the left channel signal. As a result, a signal m1, which is a main signal, can be obtained. The signal subtracting section 204 subtracts the right channel signal from the left channel signal. As a result, a signal m2 that is a noise reference signal can be obtained.

  The configuration in FIG. 19 assumes that the present invention is applied to a device using a one-point stereo microphone, such as a video movie. For example, in this device, stereo sound pickup is usually performed, and when only the front direction is emphasized as the target sound, directivity re-synthesis as described below may be performed.

  In a normal one-point stereo microphone, the left and right microphones are set in consideration of the sound image localization at the time of reproduction so that the phases of sounds coming from the center (the front direction shown in FIG. 19) are the same in the left and right microphone units. The same amplitude and phase characteristics of the units are used. As described above, the directivity angles of the microphone units 3 and 4 are set to the same angle for the left and right microphone units. Therefore, by adding the right channel signal and the left channel signal in the signal adding unit 205, a signal m1 having directivity in the front direction can be obtained. Further, the signal subtracting section 204 subtracts the right channel signal from the left channel signal to obtain a signal m2 having a directional blind spot in the front direction. As described above, signal m1 and signal m2 generated by directivity re-combining section 200 are the same as signal m1 and signal m2 in the first embodiment, respectively. Therefore, it is possible to perform noise suppression processing and reflection characteristic distortion correction processing using the signal m1 and the signal m2.

  As described above, according to the tenth embodiment, it is possible to emphasize the sound in the target sound direction by using the signal output from the one-point stereo microphone. Therefore, it becomes possible for a device having a one-point stereo microphone to function as, for example, a zoom microphone. Further, in the tenth embodiment, since directivity is re-synthesized based on a stereo signal, it can be applied to multi-channel sound collection in which a stereo signal and a signal in the front direction are obtained at the same time. In addition, even if the stereo microphone is an analog circuit, the same effect as described above can be obtained.

(Embodiment 11)
Next, a microphone device according to Embodiment 11 will be described with reference to FIG. The eleventh embodiment aims at obtaining a main signal and a noise reference signal required for the noise suppression processing of the present invention in a device that generates a stereo signal.

  FIG. 20 is a diagram illustrating a configuration of the microphone device according to the eleventh embodiment. In FIG. 20, the microphone device includes a fifth microphone unit 5, a sixth microphone unit 6, a directivity synthesis unit 500, and a directivity re-synthesis unit 200. Each of the microphone units 5 and 6 is an omnidirectional microphone unit having the same characteristics. The arrangement positions of the microphone units 5 and 6 are the same as the arrangement shown in FIG. Directional synthesis section 500 receives signals output from microphone units 5 and 6 as inputs, and outputs right channel signal Rch and left channel signal Lch. The directivity re-synthesis unit 200 receives the right channel signal Rch and the left channel signal Lch as inputs, and is a signal m1 which is a main signal having sensitivity in the target sound direction and a noise reference signal having a directional blind spot in the target sound direction. And outputs a signal m2. Note that the target sound direction can be set to a direction other than the front direction.

  Also, in FIG. 20, the directivity re-synthesis unit 200 includes an inverse directivity synthesis unit 250 and a directivity synthesis unit 100. Reverse directivity synthesis section 250 receives as input signals (right channel signal Rch and left channel signal Lch) output from directivity synthesis section 500. The reverse directivity synthesis unit 250 generates an omnidirectional signal from the right channel signal Rch and the left channel signal Lch. Directivity combining section 100 is the same as that shown in the fifth embodiment. However, here, it is assumed that the angle setting unit 160 is not provided. In FIG. 20, the directivity synthesis unit 100 has the configuration shown in FIG. 18A. However, the directivity synthesis unit 100 has the configuration shown in FIGS. 15, 16A, and 17A. There may be.

  In the eleventh embodiment, the stereo signals (right channel signal Rch and left channel signal Lch) obtained by directivity synthesis section 500 are re-converted into signals output from microphone units 5 and 6 by reverse directivity synthesis section 250. Is converted. That is, the stereo signal is reconverted into two omnidirectional signals. Further, the omnidirectional signal obtained by the reconversion is converted by the directivity synthesis unit 100 into a main signal for detecting a target sound arriving from a predetermined direction and a noise reference signal.

ここで、左辺のｘ１およびｘ２は第５および第６のマイクロホンユニット５および６からそれぞれ出力される信号であり、右辺のＲｃｈおよびＬｃｈが指向性合成部５００から出力されるステレオ信号である。なお、指向性合成部５００については、一般に用いられている指向性合成のための構成であるので、詳細説明は省略する。なお、式（１）において、１／（１−Ｈτ４（ω））の部分は、６ｄｂ／ｏｃｔの周波数特性補正項になる。実際のマイクロホン装置では補正が行われるが、指向特性とは別に考えられるのでここでは無視している。指向性合成部５００によって得られたステレオ信号（信号Ｒｃｈおよび信号Ｌｃｈ）をマイクロホンユニットから出力された信号（信号ｘ１および信号ｘ２）に戻すには、式（１）の左辺第２項の行列の逆行列を両辺の左側から掛ければよく、いわゆる逆フィルタによって実現することができる。このことを数式で表現すると（２）、（３）のようになる。

従って、信号Ｒｃｈおよび信号Ｌｃｈに式（３）の処理を行うことによって、逆指向性合成を実現することができる。図２０に示す逆指向性合成部２５０は、式（３）を図示したものである。指向性合成部１００は、このようにして得られた信号ｘ１および信号ｘ２から、目的音方向に感度を持つ主信号ｍ１と、目的音方向に指向性死角を持つ雑音参照信号ｍ２とを生成する。 Here, the directivity synthesizing unit 500 for outputting a stereo signal includes a first signal delay unit 501, a first signal subtraction unit 521, a second signal delay unit 502, and a second signal subtraction unit. 522. First signal delay section 501 delays and outputs a signal output from sixth microphone unit 6. The first signal subtraction unit 521 subtracts the signal output from the first signal delay unit 501 from the signal output from the fifth microphone unit 5, and outputs a signal Rch obtained as a result of the subtraction. Second signal delay section 502 delays the signal output from fifth microphone unit 5 and outputs the delayed signal. The second signal subtraction unit 522 subtracts the signal output from the second signal delay unit 502 from the signal output from the sixth microphone unit 6, and outputs a signal Lch obtained as a result of the subtraction. The operation of the directivity synthesis unit 500 described above is expressed by a mathematical expression as follows.

Here, x1 and x2 on the left side are signals output from the fifth and sixth microphone units 5 and 6, respectively, and Rch and Lch on the right side are stereo signals output from the directivity combining unit 500. It should be noted that the directivity synthesis unit 500 has a configuration for directivity synthesis that is generally used, and thus detailed description is omitted. In equation (1), the portion of 1 / (1−Hτ4 (ω)) is a frequency characteristic correction term of 6 db / oct. Correction is performed in an actual microphone device, but is ignored here because it can be considered separately from the directional characteristics. To return the stereo signals (signal Rch and signal Lch) obtained by the directivity synthesis unit 500 to the signals (signal x1 and signal x2) output from the microphone unit, the matrix of the second term of the second term on the left side of Expression (1) What is necessary is just to multiply the inverse matrix from the left side of both sides, and it can be realized by a so-called inverse filter. This can be expressed by mathematical expressions as shown in (2) and (3).

Therefore, by performing the processing of Expression (3) on the signal Rch and the signal Lch, it is possible to realize reverse directivity synthesis. The backward directivity synthesis unit 250 shown in FIG. 20 illustrates Equation (3). The directivity synthesis unit 100 generates a main signal m1 having sensitivity in the target sound direction and a noise reference signal m2 having directivity blind spot in the target sound direction from the signals x1 and x2 obtained in this manner. .

  As described above, Embodiment 11 utilizes a signal output from a one-point stereo microphone. In this case, the same effect as in the tenth embodiment can be obtained. That is, it is possible to enhance the target sound arriving from the front direction and correct the frequency distortion due to reflection. Further, the eleventh embodiment can deal with a target sound arriving from an arbitrary direction.

  The eleventh embodiment is particularly effective when a signal output from the microphone unit is not obtained and only a stereo signal is available. In other words, according to the eleventh embodiment, a configuration for obtaining the main signal of the target sound and the ideal noise reference signal can be realized even in a device that generates a stereo signal.

  FIG. 21 is a diagram illustrating an application example of the eleventh embodiment. FIG. 21 is a diagram showing a system including an audio recording device 801 and an audio reproducing device 802. The audio recording device 801 includes fifth and sixth microphone units 5 and 6, and a directivity synthesis unit 500. The recording unit 803 is a recording medium that is detachable from the audio recording device 801 and the audio reproducing device 802. The audio reproducing device 802 includes the directivity re-synthesizing unit 200. Although not shown, the audio reproducing device 802 has the configuration of any one of the microphone devices according to the first to fourth embodiments.

  21, a signal Rch and a signal Lch are recorded in a recording unit 803 of the audio recording device 801. Thus, the audio information has been recorded in the recording unit 803. When the recording unit 803 in which the audio information is recorded is attached to the audio reproducing device 802, the audio reproducing device 802 reads the information recorded in the recording unit 803. Specifically, the signal Rch and the signal Lch are read by the directivity re-combining unit 200. The directional resynthesis unit 200 generates a main signal and a noise reference signal from the read signal Rch and signal Lch. By using the main signal and the noise reference signal, it is possible to perform noise suppression processing on the target sound.

  As described above, the configuration of the eleventh embodiment can be realized even when the audio recording device 801 and the audio reproducing device 802 are separate bodies. That is, it is also possible to perform noise suppression processing on a signal once recorded in the recording unit 803 such as a video movie at the time of reproduction.

  FIG. 22 is a diagram illustrating an application example of the audio reproduction device illustrated in FIG. 21. 22, the sound reproducing device 802 includes an image display unit 900 and an angle setting unit 160 in addition to the configuration described in FIG. That is, the sound reproducing device 802 shown in FIG. 22 has an image display function, and is realized by, for example, a digital video camera or the like.

  In FIG. 22, the recording unit 803 records image information to be displayed on the image display unit in addition to the audio information described in FIG. The audio information and the image information are information related to each other, such as image (video) and audio information recorded simultaneously by a digital video camera. The audio information and the image information are simultaneously reproduced in the audio reproducing device 802. Here, during reproduction of the audio information and the image information, the user uses the angle setting unit 160 to specify an angle. At this time, the user determines the angle while viewing the image displayed on the image display unit. For example, if the subject is displayed at the center of the screen of the image display unit, the user indicates an angle indicating a direction corresponding to the center of the screen (that is, the front direction). Thereby, the user can extract and hear the sound coming from the front direction as the target sound.

  In the other embodiments, the following configuration is also conceivable. FIG. 23 is a diagram illustrating a part of a configuration of a microphone device according to another embodiment. 23, the fifth microphone unit 5, the sixth microphone unit 6, and the directivity synthesis unit 500 have the same configuration as that shown in FIG. The directional resynthesis unit 200 has the same configuration as that shown in FIG. According to the configuration shown in FIG. 23, the same effect as above can be obtained. Note that any of the configurations of Embodiments 1 to 4 is used for the configuration after obtaining the signal m1 and the signal m2.

  As described above, according to the present invention, for a directional microphone output directed to a target sound direction, stationary and non-stationary noises are suppressed in directions other than the target sound direction. Can be obtained. At the same time, it is possible to remove the influence of the reflected wave on the frequency characteristics of the microphone device. As described above, from the effect, it is possible to simultaneously suppress both the noise that is additive noise and the reflected wave that is multiplicative noise, and achieve a high S / N and always flat microphone frequency characteristic without being affected by the sound field. Can be realized. In the noise suppression processing unit, by realizing a configuration in which the processing delay is reduced, it is possible to apply the present invention to a loudspeaker or a call in which a large delay is not allowed. In addition, by combining directivity synthesis, reverse directivity synthesis, directivity resynthesis, and the like as preprocessing, sounds in various directions can be extracted, and the same effect on the reproduction device side can be obtained.

  As described above, the microphone device and the reproducing device according to the present invention can be operated stably even under a plurality of noises in an actual use environment, and can be used for the purpose of realizing a high S / N.

FIG. 2 is a block diagram showing a configuration of the microphone device according to the first embodiment. The figure which shows the structure of the determination part shown in FIG. The figure which shows the example of the state of audio | voice detection when the direction of the dominant sound is (theta) 1- (theta) 3 direction. The figure which shows the example of a structure of the noise suppression filter coefficient calculation part 40. The figure which shows the example of a structure of the time-varying coefficient filter part 50. The figure which shows the other example of a structure of the time-varying coefficient filter part 50. The figure which shows the specific example of each signal shown in FIG. FIG. 4 is a block diagram showing a configuration of a microphone device according to Embodiment 2. The figure explaining the difference of the internal state of the microphone device when there is a reflective object and when there is no reflective object. FIG. 9 is a block diagram showing a configuration of a microphone device according to Embodiment 3. FIG. 9 is a block diagram showing another configuration of the microphone device according to the third embodiment. FIG. 9 is a block diagram showing a configuration of a microphone device according to Embodiment 4. Diagram showing directivity pattern of microphone device FIG. 14 is a diagram showing a part of the configuration of the microphone device according to the fifth embodiment. FIG. 14 shows a part of the configuration of the microphone device according to the sixth embodiment. FIG. 14 shows a part of the configuration of the microphone device according to the seventh embodiment. FIG. 14 shows a part of the configuration of the microphone device according to the eighth embodiment. FIG. 14 shows a part of the configuration of the microphone device according to the ninth embodiment. FIG. 14 shows a part of the configuration of the microphone device according to the tenth embodiment. FIG. 14 shows a configuration of a microphone device according to Embodiment 11. FIG. 27 shows an application example of the eleventh embodiment. The figure which shows the example of application of the audio | voice reproduction apparatus shown in FIG. FIG. 9 is a diagram illustrating a part of a configuration of a microphone device according to another embodiment. FIG. 2 is a diagram illustrating a configuration of a microphone device of Conventional Example 1. FIG. 3 is a diagram illustrating a configuration of a microphone device of Conventional Example 2; FIG. 3 is a diagram showing a configuration of a microphone device of Conventional Example 3;

Explanation of reference numerals

REFERENCE SIGNS LIST 1 first microphone unit 2 second microphone unit 3 third microphone unit 4 fourth microphone unit 10 determination unit 20 adaptive filter unit 30 signal subtraction unit 40 noise suppression filter coefficient calculation unit 50 time-varying coefficient filter unit 60 reflection Information calculation unit 70 Reflection correction unit 90 Detection threshold setting unit

Claims (19)

A microphone device for detecting a target sound coming from a target sound direction,
A main signal indicating the result of detection with sensitivity to the target sound direction, and a noise reference signal indicating the result of detecting sound arriving from a direction other than the target sound direction with higher sensitivity than the target sound. A signal generation unit to generate;
A determination unit that determines whether a level ratio indicating a ratio of a signal level of the main signal to a signal level of the noise reference signal generated by the signal generation unit is larger than a predetermined value,
By filtering the main signal generated by the signal generation unit with an adaptive filter, a signal indicating the signal component of the target sound included in the noise reference signal generated by the signal generation unit is generated, and the determination unit Only when it is determined that the level ratio is larger than a predetermined value, an adaptive filter unit that learns a filter coefficient,
From the noise reference signal generated by the signal generation unit, a subtraction unit that subtracts the signal generated by the adaptive filter unit,
A microphone device comprising: a noise suppression unit that suppresses a signal component of noise included in a main signal using the main signal and a noise reference signal after the subtraction by the subtraction unit. A microphone device for detecting a target sound coming from a target sound direction,
A main signal indicating the result of detection with sensitivity to the target sound direction, and a noise reference signal indicating the result of detecting sound arriving from a direction other than the target sound direction with higher sensitivity than the target sound. A signal generation unit to generate;
A determination unit that determines whether a level ratio indicating a ratio of a signal level of the main signal to a signal level of the noise reference signal generated by the signal generation unit is larger than a predetermined value,
By filtering the main signal generated by the signal generation unit with an adaptive filter, a signal indicating the signal component of the target sound included in the noise reference signal generated by the signal generation unit is generated, and the determination unit Only when it is determined that the level ratio is larger than a predetermined value, an adaptive filter unit that learns a filter coefficient,
From the noise reference signal generated by the signal generation unit, a subtraction unit that subtracts the signal generated by the adaptive filter unit,
Based on the filter coefficient of the adaptive filter unit, a reflection information calculation unit that calculates information about the arrival time difference between the direct wave and the reflected wave of the target sound,
A microphone device, comprising: a reflection correction unit configured to correct distortion of a frequency characteristic generated in a main signal due to a reflected wave of a target sound based on information calculated by the reflection information calculation unit. The signal generator,
A first microphone unit in which a directional principal axis is arranged to face a target sound direction;
The microphone device according to claim 1, further comprising: a second microphone unit arranged such that the blind spot direction of the directivity is directed to the target sound direction.   A signal delay unit that is provided between an output terminal of the noise reference signal in the signal generation unit and the subtraction unit and delays the noise reference signal so as to satisfy a convergence condition of an adaptive filter of the adaptive filter unit. The microphone device according to claim 1.   3. The microphone device according to claim 1, wherein the predetermined value is changeable. The signal generator,
A first microphone unit;
A second microphone unit having the same characteristics as the first microphone unit;
A delay unit that delays and outputs a signal output from the first microphone unit by a predetermined delay amount;
An amplification unit that amplifies the signal output from the delay unit,
A first subtraction unit that generates a main signal by subtracting a signal amplified by the amplification unit from a signal output from the second microphone unit;
A second subtraction unit that generates a noise reference signal by subtracting a signal output from the delay unit from a signal output from the second microphone unit,
A straight line connecting the position of the first microphone unit and the position of the second microphone unit is orthogonal to a straight line directed to a target sound direction,
The predetermined amount of delay is set so that the noise reference signal contains more components of the sound arriving from directions other than the target sound direction than components of the target sound,
The microphone device according to claim 1, wherein an amplification factor in the amplification unit is set such that a difference occurs in sensitivity of a target sound between the main signal and the noise reference signal.   The microphone device according to claim 6, further comprising a setting unit that changes a predetermined delay amount set in the delay unit. The signal generator,
A first microphone unit;
A second microphone unit having the same characteristics as the first microphone unit;
A main signal is generated based on each signal output from the first and second microphone units so as to have sensitivity in a target sound direction, and a noise signal is generated so that sensitivity in the target sound direction is minimized. The microphone device according to claim 1, further comprising: a synthesis unit configured to generate The signal generator,
A first microphone unit;
A second microphone unit having a directivity main axis directed in a direction different from that of the first microphone unit,
A signal adding unit that generates a main signal by adding a signal output from the first microphone unit and a signal output from the second microphone unit;
And a signal subtracting section that generates a noise reference signal by subtracting one of the signal output from the first microphone unit and the signal output from the second microphone unit from the other. 3. The microphone device according to 1 or 2. The signal generator,
A first microphone unit;
A second microphone unit having the same characteristics as the first microphone unit;
A stereo signal generation unit that generates a stereo signal including a right channel signal and a left channel signal based on the first and second microphone units;
An inverse synthesizing unit that generates each signal output from each microphone unit based on the stereo signal,
Based on each signal generated by the inverse synthesis unit, a main signal indicating the result of detection with sensitivity to the target sound direction, and a sound arriving from a direction other than the target sound direction, The microphone device according to claim 1, further comprising: a synthesis unit configured to generate a noise reference signal indicating a result of detection with high sensitivity. The signal generator,
A first microphone unit;
A second microphone unit having the same characteristics as the first microphone unit;
A stereo signal generation unit that generates a stereo signal including a right channel signal and a left channel signal based on the first and second microphone units;
A signal addition unit that generates a main signal by adding the right channel signal and the left channel signal of the stereo signal,
The microphone device according to claim 1, further comprising: a signal subtraction unit configured to generate a noise reference signal by subtracting one of the right channel signal and the left channel signal of the stereo signal from the other. Based on the filter coefficient of the adaptive filter unit, a reflection information calculation unit that calculates information about the arrival time difference between the direct wave and the reflected wave of the target sound,
A reflection correction unit that corrects a distortion of a frequency characteristic generated in a main signal by a reflected wave of the target sound based on the information calculated by the reflection information calculation unit,
2. The noise suppression unit according to claim 1, wherein the noise suppression unit suppresses a signal component of noise included in the main signal using the main signal corrected by the reflection correction unit and a noise reference signal subtracted by the subtraction unit. 3. A microphone device as described. The noise suppression unit includes:
A noise suppression filter coefficient calculation unit that calculates a filter coefficient of a noise suppression filter for suppressing a signal component of noise from the main signal based on the main signal and the noise reference signal after the subtraction by the subtraction unit;
The microphone device according to claim 1, further comprising: a time-varying coefficient filter unit that performs filtering on the main signal by the noise suppression filter by reflecting the filter coefficient calculated by the noise suppression filter coefficient calculation unit. The noise suppression filter coefficient calculation unit,
A first frequency analyzer for calculating a power spectrum of the main signal;
A second frequency analysis unit that calculates a power spectrum of the noise reference signal after the subtraction by the subtraction unit;
Only when the determining unit determines that the level ratio is smaller than a predetermined value, the power spectrum calculated by the first frequency analyzing unit and the power spectrum calculated by the second frequency analyzing unit. A power spectrum ratio calculation unit that calculates a time average of a power spectrum ratio with a spectrum,
A time average of the power spectrum ratio calculated by the power spectrum ratio calculation unit, and a multiplication unit that multiplies the power spectrum calculated by the second frequency analysis unit;
2. The microphone according to claim 1, further comprising: a coefficient calculator that calculates a filter coefficient of the noise suppression filter based on a power spectrum calculated by the first frequency analyzer and a multiplication result by the multiplier. 3. apparatus. A microphone device for detecting a target sound coming from a target sound direction,
A first microphone unit;
A second microphone unit having a directivity main axis directed in a direction different from that of the first microphone unit,
A signal adding unit that generates a main signal by adding a signal output from the first microphone unit and a signal output from the second microphone unit;
A signal subtraction unit that generates a noise reference signal by subtracting the signal output from the first microphone unit and the signal output from the second microphone unit from one of the other;
A microphone device comprising: a noise suppression unit configured to suppress a signal component of noise included in a main signal using the main signal and the noise reference signal. A microphone device for detecting a target sound coming from a target sound direction,
A first microphone unit;
A second microphone unit having the same characteristics as the first microphone unit;
A stereo signal generation unit that generates a stereo signal including a right channel signal and a left channel signal based on the first and second microphone units;
An inverse synthesizing unit that generates each signal output from each microphone unit based on the stereo signal,
Based on each signal generated by the inverse synthesis unit, a main signal indicating the result of detection with sensitivity to the target sound direction, and a sound arriving from a direction other than the target sound direction, A synthesis unit that generates a noise reference signal indicating a result of detection with high sensitivity,
A microphone device comprising: a noise suppression unit configured to suppress a signal component of noise included in a main signal using the main signal and the noise reference signal. A microphone device for detecting a target sound coming from a target sound direction,
A first microphone unit;
A second microphone unit having the same characteristics as the first microphone unit;
A stereo signal generation unit that generates a stereo signal including a right channel signal and a left channel signal based on the first and second microphone units;
A signal addition unit that generates a main signal by adding the right channel signal and the left channel signal of the stereo signal,
A signal subtraction unit that generates a noise reference signal by subtracting the other from one of the right channel signal and the left channel signal of the stereo signal,
A microphone device comprising: a noise suppression unit configured to suppress a signal component of noise included in a main signal using the main signal and the noise reference signal. An audio recording unit that records audio signals of at least two types of channels;
Based on the audio signal recorded in the recording unit, a main signal indicating the result of detection with sensitivity to the target sound direction, and a sound arriving from a direction other than the target sound direction, A signal generation unit that generates a noise reference signal indicating a result of detection with high sensitivity,
A determination unit that determines whether a level ratio indicating a ratio of a signal level of the main signal to a signal level of the noise reference signal generated by the signal generation unit is larger than a predetermined value,
By filtering the main signal generated by the signal generation unit with an adaptive filter, a signal indicating the signal component of the target sound included in the noise reference signal generated by the signal generation unit is generated, and the determination unit Only when it is determined that the level ratio is larger than a predetermined value, an adaptive filter unit that learns a filter coefficient,
From the noise reference signal generated by the signal generation unit, a subtraction unit that subtracts the signal generated by the adaptive filter unit,
Using the main signal and a noise reference signal after the subtraction by the subtraction unit, a noise suppression unit that suppresses a signal component of noise included in the main signal,
A reproducing unit configured to reproduce a main signal in which a noise signal component is suppressed by the noise suppressing unit. A video recording unit that records a video signal related to the audio signal recorded in the audio recording unit,
A video reproduction unit that reproduces a video signal recorded in the video recording unit,
A direction receiving unit that receives an input of a direction in which the sound should be emphasized from the user,
19. The audio reproducing device according to claim 18, wherein the signal generation unit generates a main signal and a noise reference signal with a direction received by the direction reception unit as a target sound direction.

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