JP6624255B1 - Sound pickup device, program and method - Google Patents

Abstract

PROBLEM TO BE SOLVED: To provide a sound collecting device, a program and a method for collecting a target area sound with less distortion. The present invention relates to a sound collection device. Then, the sound collection device of the present invention obtains the beamformer output of each microphone array based on the input signals input from the plurality of microphone arrays, and uses the obtained beamformer output to set the target area as a sound source. A target area sound extracting means for extracting a target area sound, and for each frequency, a component of an input signal of each microphone constituting each microphone array is compared, and a component of an input signal of any one of the microphones is compared with a mixed signal. Selecting means for selecting a component as a component, a signal mixing means for mixing a target area sound component extracted by the target area sound extracting means with a mixed signal composed of components selected for each frequency by the selecting means, And an output means for outputting a mixed signal after mixing. [Selection diagram] Fig. 1

Description

  The present invention relates to a sound collection device, a program, and a method, and can be applied to, for example, a system that emphasizes sound in a specific area and suppresses sound in other areas.

  In an environment where a plurality of sound sources are present, as a technique for separating and collecting only sound in a specific direction, there is a beam former (hereinafter also referred to as “BF”) using a microphone array. BF is a technique for forming directivity by using a time difference between signals reaching each microphone (see Non-Patent Document 1).

  Conventionally, BFs are broadly classified into two types: an addition type and a subtraction type. In particular, the subtraction type BF has an advantage that directivity can be formed with a smaller number of microphones than the addition type BF.

  FIG. 4 is a block diagram showing a configuration of the subtraction type BF 200 when the number of microphones M is two.

  FIG. 5 is an explanatory diagram showing an example of a directional filter formed by a subtractive BF 200 using two microphones M1 and M2.

  The subtraction type BF 200 first calculates the time difference between the signals that arrive at the microphones M1 and M2 of the sound (hereinafter, referred to as “target sound”) existing in the target direction by the delay unit 210, and adds the delay to the target. Adjust the sound phase. The above-mentioned time difference can be calculated by the following equation (1).

Here, d is the distance between the microphones M1 and M2, c is the speed of sound, and τ _i is the amount of delay. The theta _L is the angle from the vertical direction to the target direction against the line connecting the microphones M (M1, M2).

Also, here, with respect to the center of the blind spot is a microphone M1 M2, when present in the direction of the microphone M1, delayer 210 performs delay processing on an input signal _x 1 of the microphone M1 (t). Thereafter, the subtraction type BF 200 performs processing (subtraction processing) according to the following equation (2).

The processing of the subtraction type BF 200 can be similarly performed in the frequency domain, and in that case, the equation (2) is changed to the following (3).

Here, when θ _L = ± π / 2, the directivity formed by the subtraction type BF 200 is a cardioid type single directivity as shown in FIG. Further, in the case of “θ _L = 0, π”, the directivity formed by the subtraction type BF 200 is a figure-eight bidirectional directivity as shown in FIG. 5B.

  Hereinafter, a filter that forms unidirectionality from an input signal is referred to as a “unidirectionality filter”, and a filter that forms bidirectionality is referred to as a bidirectional filter.

  In addition, the subtractor 220 can form a strong directivity in a bidirectional blind spot by using a spectral subtraction method (hereinafter, also simply referred to as “SS”). The directivity by the SS is formed in all frequencies or a designated frequency band according to the following equation (4).

In the following equation (4), and using the input signal X ₁ microphone M1, but it is possible to obtain the same effect input signal X ₂ microphones M2. Here, β is a coefficient for adjusting the strength of SS. When the value becomes negative at the time of subtraction, the subtractor 220 performs flooring processing of replacing it with 0 or a value obtained by reducing the original value. In the processing method of the subtraction type BF 200 as described above, a sound existing in a direction other than the target direction (hereinafter, referred to as a “non-target sound”) is extracted due to the bidirectional characteristics, and an amplitude spectrum of the extracted non-target sound is input. By subtracting from the amplitude spectrum of the signal, the target sound can be emphasized.

  When it is desired to collect only sounds existing in a certain area (hereinafter, referred to as “target area sound”), sound of a sound source existing around the area (hereinafter, “non-target sound”) is obtained only by using the subtraction type BF. (Referred to as “target area sound”). Therefore, in Patent Document 1, a method of collecting a target area sound by using a plurality of microphone arrays, directing the directivity to the target area from different directions, and intersecting the directivity at the target area (hereinafter referred to as “area collection”). Sound). In the area sound pickup, first, the ratio of the amplitude spectrum of the target area sound included in the BF output of each microphone array is estimated, and the ratio is used as a correction coefficient.

For example, when two microphone arrays are used, the correction coefficient of the target area sound amplitude spectrum is calculated by a combination of the following equations (5) and (6) or a combination of the following equations (7) and (8). Can be calculated. Here, Y _1k (n) is the amplitude spectrum of the BF output of the first microphone array, Y _2k (n) is the amplitude spectrum of the BF output of the second microphone array, and N is the total number of frequency bins. And k is the frequency. Here, α ₁ (n) and α ₂ (n) are amplitude spectrum correction coefficients for each BF output. Further, here, mode represents the mode, and median represents the median.

By the above processing, the subtractor 220 calculates the correction coefficients α ₁ (n) and α ₂ (n), corrects each BF output by the obtained correction coefficients, and performs SS to obtain the non-existence in the target area direction. Extract the target area sound. Further, the subtracter 220 can extract the target area sound by applying the extracted non-target area sound from the output of each BF.

When extracting the non-target area sound N ₁ (n) existing in the target area direction viewed from the first microphone array, the subtraction type BF 200, for example, as shown in Expression (9), outputs the BF output of the first microphone array. SS obtained by multiplying the BF output Y ₂ (n) of the second microphone array by the amplitude spectrum correction coefficient α ₂ from Y ₁ (n). Similarly, the subtraction type BF ₂₀₀ extracts the non-target area sound N ₂ (n) existing in the target area direction viewed from the second microphone array according to the following equation (10).

Thereafter, the subtraction type BF 200 extracts the target area sound by SS the non-target area sound from each BF output according to the following equation (11) or (12). Expression (11) below shows a process for extracting a target area sound based on the first microphone array. The following equation (12) shows a process for extracting a target area sound based on the second microphone array. Here, γ ₁ (n) and γ ₂ (n) are coefficients for changing the strength at the time of SS.

  By the way, when the volume level of the background noise or the non-target area sound is large, the target area sound may be distorted or annoying unusual noise such as musical noise may occur due to SS performed at the time of extracting the target area sound.

  Therefore, in the method of Patent Document 3, the volume levels of the microphone input signal and the estimated noise are adjusted according to the loudness of the background noise and the non-target area sound, respectively, and are mixed with the extracted target area sound.

  Since the musical noise generated by the process of extracting the target area sound becomes stronger as the volume levels of the background noise and the non-target area sound become larger, the technique of Patent Document 3 discloses the volume level of the sum of the mixed input signal and the estimated noise. Are also increased in proportion to the volume levels of the background noise and the non-target area sound.

  Specifically, in the method of Patent Document 3, the volume level of the background noise is calculated from the estimated noise obtained in the process of suppressing the background noise. Further, in the method of Patent Document 3, the volume level of the non-target area sound is divided into the non-target area sound existing in the target area direction extracted in the process of enhancing the target area sound and the non-target area sound existing outside the target area direction. Calculated from the combined sound. Furthermore, in the method of Patent Document 3, the ratio between the input signal to be mixed and the estimated noise is determined from the volume levels of the estimated noise and the non-target area sound.

  When a non-target area sound exists near the target area, if the volume level of the input signal to be mixed is too high, the non-target area sound mixes with the target area sound, and it becomes difficult to determine which is the target area sound. Therefore, in the method of Patent Document 3, when the non-target area sound is loud, the volume level of the input signal to be mixed is lowered, and the volume level of the estimated noise is increased to mix. That is, in the method of Patent Document 3, when the non-target area sound does not exist or the volume level is low, the ratio of the input signal is increased, and when the volume level of the non-target area sound is high, the ratio of the estimated noise is increased. Mix.

  Thus, by using the technique of Patent Document 3, by mixing the input signal and the estimated noise with the sound of the target area, the musical noise can be masked and can be heard without discomfort like ordinary background noise. Further, according to the method of Patent Document 3, distortion of the target area sound can be corrected by the component of the target area sound included in the microphone input signal, and the sound quality can be improved.

JP 2014-72708 A JP-A-2005-195555 JP 2017-183902 A

Tadashi Asano, "Acoustic Technology Series 16: Array Signal Processing of Sound-Localization, Tracking and Separation of Sound Sources", edited by The Acoustical Society of Japan, Corona Publishing, February 25, 2011

  However, in the method of Patent Document 3, when a non-target area sound exists near the target area, the level of the input signal to be mixed is reduced, so that mixing of the non-target area sound can be suppressed. The effect of improving distortion is reduced.

  Therefore, a sound collection device, a program, and a method for collecting a target area sound with less distortion are desired.

According to a first aspect of the present invention, there is provided a sound collection apparatus which (1) acquires beamformer outputs of the respective microphone arrays based on input signals inputted from a plurality of microphone arrays, and uses the acquired beamformer outputs. A target area sound extracting means for extracting a target area sound using an area as a sound source; and (2) comparing input signal components of respective microphones constituting each of the microphone arrays for each frequency, and Selecting means for selecting a component of the input signal of the microphone as a component of the mixed signal; and (3) selecting a target area sound component extracted by the target area sound extracting means by a component selected for each frequency by the selecting means. possess a signal mixing means for mixing the mixed signal composed, and output means for outputting the mixed signal after the mixing (4) the selection means (5) the selection means, and selects as a component of each of the respective input signal mixed signal components of the most amplitude spectrum is small input signal components of the microphone constituting the microphone array.

According to a second aspect of the present invention, there is provided a sound collection program for: (1) obtaining a beamformer output of each of the microphone arrays based on input signals input from a plurality of microphone arrays; A target area sound extracting means for extracting a target area sound using the target area as a sound source, and (2) comparing the components of the input signals of the microphones constituting each of the microphone arrays for each frequency. Selecting means for selecting the component of the input signal of the microphone as a component of the mixed signal; and (3) selecting the target area sound component extracted by the target area sound extracting means for each frequency by the selecting means. Signal mixing means for mixing a mixed signal composed of the mixed components; and (4) outputting a mixed signal mixed by the selecting means. To function as output means for, (5) the selection means to select as a component of each of the respective input signal mixed signal components of the most amplitude spectrum is small input signal components of the microphone constituting the microphone array It is characterized by that.

According to a third aspect of the present invention, there is provided a sound collection program executed by a sound collection device, comprising: (1) a target area sound extraction unit, a selection unit, a signal mixing unit, and an output unit; (3) acquiring a beamformer output of each of the microphone arrays based on an input signal input from the microphone array, and extracting a target area sound having a target area as a sound source using the acquired beamformer output; The selection means, for each frequency, by comparing the components of the input signal of each microphone constituting each microphone array, to select the component of the input signal of any of the microphones as a component of the mixed signal, (4) The signal mixing unit selects the target area sound component extracted by the target area sound extraction unit for each frequency by the selection unit. Mixing the mixed signal composed of the components, (5) said output means outputs the mixed signal after said selection means are mixed, (6) the selection means, constituting each of the microphone array A component of the input signal having the smallest amplitude spectrum is selected as a component of the mixed signal from components of the input signal of each of the microphones .

  According to the present invention, it is possible to provide a sound collecting apparatus, a program, and a method for collecting a target area sound with less distortion.

FIG. 2 is a block diagram illustrating a functional configuration of the sound collection device according to the first embodiment. FIG. 5 is an explanatory diagram showing an effect of the first embodiment. It is a block diagram showing a functional configuration of a sound collection device according to a second embodiment. FIG. 11 is a block diagram showing a configuration related to a conventional subtraction type BF when the number of microphones is two. FIG. 9 is a diagram illustrating a directional characteristic formed by a conventional subtraction type BF using two microphones.

(A) First Embodiment Hereinafter, a first embodiment of a sound collection device, a program, and a method according to the present invention will be described in detail with reference to the drawings.

(A-1) Configuration of First Embodiment FIG. 1 is a block diagram showing a functional configuration of a sound collection device 100 of this embodiment.

  The sound collection device 100 performs a target area sound collection process of collecting a target area sound from a sound source in the target area by using two microphone arrays MA (MA1 and MA2).

  The microphone arrays MA1 and MA2 are arranged at any place in the space where the target area exists. The positions of the microphone arrays MA1 and MA2 with respect to the target area may be anywhere as long as the directivity overlaps only in the target area. For example, the microphone arrays MA1 and MA2 may be arranged to face each other across the target area. Each microphone array MA includes two or more microphones M, and each microphone M collects an acoustic signal. In this embodiment, a description will be given assuming that two microphones M (M1, M2) that collect sound signals are arranged in each microphone array MA. That is, each microphone array MA constitutes a 2ch microphone array. Note that the number of microphone arrays MA is not limited to two. When there are a plurality of target areas, it is necessary to arrange the number of microphone arrays MA that can cover all the areas.

  The sound collection device 100 includes a signal input unit 101, a noise suppression unit 102, a directivity forming unit 103, a delay correction unit 104, spatial coordinate data 105, a correction coefficient calculation unit 106, a target area sound extraction unit 107, and a mixed component selection unit 108. , A signal mixing unit 109, and a signal output unit 110.

  Detailed processing of each functional block constituting the sound collection device 100 will be described later.

  The sound collection device 100 may be entirely configured by hardware (for example, a dedicated chip or the like), or may be partially or entirely configured as software (program). The sound collection device 100 may be configured by installing a program (including the determination program and the sound collection program of the embodiment) in a computer having a processor and a memory, for example.

(A-2) Operation of First Embodiment Next, an operation (a sound collection method according to the embodiment) of the sound collection device 100 according to the first embodiment having the above configuration will be described.

  The signal input unit 101 converts an acoustic signal collected by each microphone array from an analog signal to a digital signal and inputs the signal. Thereafter, the time domain is transformed to the frequency domain using, for example, a fast Fourier transform.

  The noise suppression unit 102 estimates and suppresses a background noise component included in the signal acquired by the signal input unit 101. For noise suppression by the noise suppression unit 102, for example, SS, Wiener Filtering, or the like can be used.

  The directivity forming unit 103 forms directivity in the direction of the target area by the BF according to the equation (4) with respect to the signal whose background noise is suppressed by the noise suppressing unit for each microphone array.

  The delay correction unit 104 calculates and corrects a delay generated due to a difference in distance between the target area and each microphone array. The delay correction unit 104 first obtains the position of the target area and the position of each microphone array from the spatial coordinate data 105, and calculates the difference between the arrival time of the target area sound to each microphone array. Next, a delay is added so that the sound of the target area reaches all the microphone arrays simultaneously with reference to the microphone array located farthest from the target area.

  The spatial coordinate data 105 holds all target areas, each microphone array, and positional information of the microphones constituting each microphone array. The method by which the spatial coordinate data 105 holds the position information of each microphone of each microphone array, and the specific format of the position information held by the spatial coordinate data 105 are not limited, and various data formats may be applied. it can.

  The correction coefficient calculation unit 106 calculates a correction coefficient for equalizing the amplitude spectrum of the target area sound component included in each BF output according to the formulas (5) and (6) or the formulas (7) and (8).

  The target area sound extraction unit 107 SSs each BF output data corrected by the correction coefficient calculated by the correction coefficient calculation unit 106 according to the equation (9) or (10), and extracts the non-target area sound existing in the target area direction. Extract. Further, the target area sound extraction unit 107 extracts the target area sound by subjecting the extracted noise to SS from the output of each BF according to the equation (11) or (12).

The mixed component selection unit 108 compares the amplitude spectra of the input signals of the microphones (M1, M2) constituting each microphone array (MA1, MA2) for each frequency component, and determines the frequency component having the smallest amplitude spectrum as the mixed signal component. Select as When performing area sound collection using two 2-channel microphone arrays MA1 and MA2, the mixed component selection unit 108 selects the frequency component X _MIXk (n) of the mixed signal according to the equation (13).

Here, X _11k (n) and X _12k (n) are the amplitude spectra of the respective frequencies of the input signals X ₁₁ (n) and X ₁₂ (n) of the microphones M1 and M2 constituting the microphone array MA1. Here, X _21k (n) and X _22k (n) are amplitude spectra of the input signals X ₂₁ (n) and X ₂₂ (n) of the microphones M1 and M2 constituting the microphone array MA2, respectively. is there. Further, k is a frequency (identifier of a frequency component). Furthermore, in the sound collection device 100 (the mixing component selection unit 108 and the signal mixing unit 109), the frequency band (range of k) used for signal processing may be limited by setting an upper limit and a lower limit.

When X _11k (n) or X _12k (n) is selected as the mixed signal in the mixed component selection unit 108, as shown in the equation (14), the input signal component of the microphone constituting the microphone array MA1 is The averaging may be used as the mixed signal component.

The signal mixing unit 109 includes, in the target area sound component extracted by the target area sound extraction unit 107, an input signal component selected for each frequency by the mixing component selection unit 108 (an input signal component selected for each frequency). Signals; hereinafter referred to as “mixed signals”). For example, when the signal mixing unit 109 performs the area sound collection based on the microphone array MA1 according to the equation (11), the final output W _1k (n) is mixed according to the following equation (15). Here, μ _i is a parameter for adjusting the magnitude of the component of the signal to be mixed (mixed signal). μ _i may be constant at all frequencies or may be changed for each frequency.

  As described above, when restoring the phase to the mixed output signal, the signal mixing unit 109 calculates the phase information by adding and averaging the input signals of the microphones constituting the microphone array with the target area sound extraction unit as a reference. Alternatively, the input signal of any one microphone is used.

Further, signal mixing section 109 may use the phase of the input signal selected as the mixed signal. For example, when the target area sound is extracted using Expression (11), since the signal mixing unit 109 uses the microphone array MA1 as a reference, the signal averaging of the input signal components X _11k (n) and X _12k (n) is performed. Alternatively, the phase is restored to the output signal using the phase information of either X _11k (n) or X _12k (n).

  Furthermore, the signal mixing section 109 may perform the signal mixing process after restoring the phase information to the amplitude spectrum of the target area sound and the amplitude spectrum of the mixed signal. In this case, in the signal mixing section 109, the information used for the phase restoration can be separated for the target area sound and the mixed signal. For example, in the signal mixing unit 109, the target area sound is determined by adding or averaging the input signal components of the microphones forming the microphone array, which is the reference in the target area sound extracting unit, or any one of the microphones forming the microphone array. An input signal component may be used. In the signal mixing section 109, the phase of the input signal component selected as the mixed signal component may be used as the mixed signal.

  The signal output unit 110 converts the output signal processed by the signal mixing unit 109 from the frequency domain to the time domain, and outputs the converted signal.

  As described above, in the first embodiment, as a mixed signal, the amplitude spectra of the input signals of all microphones used for area sound pickup are compared for each frequency, and the frequency component having the smallest amplitude spectrum is selected. Furthermore, in the first embodiment, when sound is collected in a specific area, it is desirable that the microphone array be installed around the sound collection area.

(A-3) Effects of First Embodiment According to the first embodiment, the following effects can be obtained.

  In the sound collection device 100 of the first embodiment, for each frequency component of the input signal of each microphone array, the one with the smallest amplitude spectrum is selected as the mixed signal component, so that the non-target area sound is close to the target area. Even in the case where the target area sound exists, it is possible to suppress the mixing of the non-target area sound after mixing, and to improve the distortion of the target area sound.

  Here, for example, in the first embodiment, it is assumed that the distance from each microphone array to the center of the sound collection area is equal. Further, for example, in the first embodiment, it is assumed that the target area sound is input to all the microphones constituting each microphone array at the same volume (see FIG. 2A). On the other hand, the position where the non-target area sound exists has a different distance from each microphone array. Therefore, the volume of the non-target area sound included in the signal of each microphone array has a different magnitude due to the distance attenuation. Also, in each microphone constituting one microphone array, if the non-target area sound exists other than in front of the microphone array, the distance between the non-target area sound and each microphone is different, so that a difference occurs in the sound volume (FIG. 2). (B)). That is, the input signal of the microphone located farthest from the non-target area sound has the smallest non-target area sound included. Therefore, since the target area sound is included in the signals of all microphones at the same volume, the frequency component of the input signal having the smallest amplitude spectrum among the signals of all microphones has the highest SN ratio. Therefore, in the first embodiment, even when a non-target area sound exists near the target area, the mixing of the non-target area sound after mixing is suppressed and the distortion of the target area sound is improved. Can be.

(B) Second Embodiment Hereinafter, a second embodiment of a sound collection device, a program, and a method according to the present invention will be described in detail with reference to the drawings.

(B-1) Configuration of Second Embodiment FIG. 3 is a block diagram showing a functional configuration of a sound collection device 100A according to a second embodiment. Have the same or corresponding reference numerals.

  Hereinafter, differences between the second embodiment and the first embodiment will be described.

  The sound collection device 100A of the second embodiment differs from the first embodiment in that a microphone array selection unit for each frequency 11 is added. Further, the sound collecting device 100A of the second embodiment differs from the first embodiment in that the target area sound extraction unit 107 is replaced with the target area sound extraction unit 107A. Note that the second embodiment is different from the first embodiment in that the processing result of the mixed component selection unit 108 is supplied to the frequency-specific microphone array selection unit 111 and the target area sound extraction unit 107A.

  Detailed processing of each functional block configuring the sound collection device 100A will be described later.

(B-2) Operation of Second Embodiment Next, an operation (a sound collection method according to the embodiment) of the sound collection device 100A of the second embodiment having the above-described configuration will be described in the first embodiment. The following description focuses on the differences from.

  The frequency-specific microphone array selecting unit 111 selects a microphone array to be used as a reference in the target area sound extracting unit 107 for each frequency based on the mixed signal component selected by the mixed component selecting unit 108.

Hereinafter, the input signals (components of the input components) of the microphones (M1, M2) of the microphone arrays (MA1, MA2) at a certain frequency p are represented by X _11p (n), X _12p (n), and X _21p (n, respectively). ), X _22p (n). Further, hereinafter, the input signals (components of the input components) of the microphones (M1, M2) of the microphone arrays (MA1, MA2) at the frequency q different from the frequency p are respectively represented by X _11q (n) and X _12q (n ), X _21q (n) and X _22q (n).

For example, when the input signal X _11p (n) of the microphone M1 included in the microphone array MA1 is selected as a mixed signal by the mixed component selection unit 108 at a certain frequency p, the target area sound extraction unit 107A sets the reference microphone The microphone array MA1 is selected as an array, and a target area sound (a component of the target area sound having the frequency p) is extracted according to Expression (16). Further, for example, when the input signal X _22q (n) of the microphone M2 configuring the microphone array MA2 is selected as the mixed signal by the mixed component selection unit 108 at the frequency q, the target area sound extraction unit 107A becomes a reference. The microphone array MA2 is selected as the microphone array, and a target area sound (a component of the target area sound having the frequency q) is extracted according to Expression (17).

  As described above, the target area sound extraction unit 107A selects, for each frequency, a microphone array corresponding to the input signal selected by the mixed component selection unit 108, and sets the target area based on the input signal of the selected microphone array. The sound is extracted and supplied to the subsequent stage (signal mixing unit 109).

(B-3) Effects of the Second Embodiment According to the second embodiment, the following effects can be obtained as compared with the first embodiment.

  In the second embodiment, the target area sound extraction unit 107A outputs, for each frequency (for each component), a microphone array serving as a supply source of the mixed signal selected by the mixed component selection unit 108 (the microphone array selection unit for each frequency 111). The target area sound is extracted based on the selected microphone array). Accordingly, in the sound collection device 100A of the second embodiment, the source of the mixed signal and the signal used in the target area sound extraction processing (the microphone array of the source) match, thereby further improving the distortion of the target area sound. The effect described above can be obtained.

(C) Other Embodiments The present invention is not limited to the above embodiments, and may include modified embodiments as exemplified below.

  (C-1) In the sound collection device of each of the above embodiments, the number of microphones of each microphone array MA used for sound collection is two, but the sound signal collected by using three or more microphones is The sound in the direction of the target area may be collected based on the sound. In each of the above embodiments, various existing systems can be applied to the number of microphones for each microphone array MA to be applied and the system for collecting sound in the target sound direction.

  100: sound collecting device, M1, M2: microphone, MA1, MA2: microphone array, 101: signal input unit, 102: noise suppression unit, 103: directivity forming unit, 104: delay correction unit, 105: spatial coordinate data, 106: correction coefficient calculating unit, 107: target area sound extracting unit, 108: mixed component selecting unit, 109: signal mixing unit, 110: signal output unit.

Claims (5)

A target area sound for acquiring a beamformer output of each of the microphone arrays based on input signals input from a plurality of microphone arrays, and extracting a target area sound having a target area as a sound source using the acquired beamformer outputs. Extraction means;
Selecting means for comparing the components of the input signals of the microphones constituting each of the microphone arrays for each frequency, and selecting the component of the input signal of any one of the microphones as a component of the mixed signal;
A signal mixing unit that mixes a target area sound component extracted by the target area sound extraction unit with a mixed signal formed by a component selected for each frequency by the selection unit;
Output means for outputting a mixed signal after the selection means has been mixed ,
The selection means selects a component of an input signal having the smallest amplitude spectrum from components of an input signal of each of the microphones constituting each of the microphone arrays as a component of a mixed signal. Sound device. 2. The sound pickup device according to claim 1, wherein the target area sound extraction unit performs a process of extracting a target area sound based on a beamformer output of any one of the microphone arrays. 3. 3. The sound pickup according to claim 2 , wherein the target area sound extracting unit performs a process of extracting a target area sound based on a beamformer output of the microphone array related to the input signal selected by the selecting unit. apparatus. Computer
A target area sound for acquiring a beamformer output of each of the microphone arrays based on input signals input from a plurality of microphone arrays, and extracting a target area sound having a target area as a sound source using the acquired beamformer outputs. Extraction means;
Selecting means for comparing the components of the input signals of the microphones constituting each of the microphone arrays for each frequency, and selecting the component of the input signal of any one of the microphones as a component of the mixed signal;
A signal mixing unit that mixes a target area sound component extracted by the target area sound extraction unit with a mixed signal formed by a component selected for each frequency by the selection unit;
Said selecting means to function as an output means for outputting the mixed signal after the mixing is,
The sound collecting program according to claim 1, wherein said selecting means selects a component of an input signal having the smallest amplitude spectrum from a component of an input signal of each of said microphones constituting each of said microphone arrays as a component of a mixed signal . In the sound pickup method performed by the sound pickup device,
Target area sound extraction means, selection means, signal mixing means, output means,
The target area sound extracting means obtains a beamformer output of each of the microphone arrays based on input signals input from a plurality of microphone arrays, and uses the obtained beamformer outputs to set a target area as a sound source. Extract the area sound,
The selection means, for each frequency, by comparing the components of the input signal of each microphone constituting each microphone array, to select the component of the input signal of any of the microphones as a component of the mixed signal,
The signal mixing unit mixes a target area sound component extracted by the target area sound extraction unit with a mixed signal formed by a component selected for each frequency by the selection unit,
The output means outputs a mixed signal mixed by the selection means ,
The sound collection method according to claim 1, wherein the selecting means selects, as a mixed signal component, an input signal component having the smallest amplitude spectrum from input signal components of each of the microphones included in each of the microphone arrays .

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