JP2020036139A - Sound pickup device, program and method - Google Patents

Abstract

To provide a sound pickup device, a program and a method capable of picking up a target area sound with less distortion.SOLUTION: The present invention relates to a sound pickup device having target area sound extraction means for acquiring the beam former outputs of respective microphone arrays, on the basis of input signals from multiple microphone arrays, and extracting a target area sound from a target area, as the sound source, by using the beam former outputs thus acquired, selection means for selecting the input signal of any microphone array as a mixed signal, on the basis of comparison results of the average amplitude spectrum of the input signals of respective microphone arrays, signal mixture means for mixing, as the mixed signal, the input signal of a microphone array, selected by the selection means, to the target area sound extracted by the target area sound extraction means, and output means for outputting the signal mixed by the selection means.SELECTED DRAWING: Figure 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 exist, 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 roughly classified into two types, namely, 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 illustrating 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 subtraction type 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”) present 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 sound speed, and τ _i is the delay amount. 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, in which case the equation (2) is changed to the following equation (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. In addition, 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.

  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 characteristic, 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 specific area (hereinafter, referred to as “target area sound”), the sound of a sound source existing around that area (hereinafter, “non (Referred to as “target area sound”). Therefore, in Japanese Patent Application Laid-Open No. H11-163, a method of collecting a target area sound by using a plurality of microphone arrays and directing the directivity from different directions to the target area so that the directivity intersects the target area is described. Sound "). In the area sound collection, 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. Furthermore, here, mode represents the mode, and median represents the median.

By the above processing, the subtracter 220 calculates the correction coefficients α ₁ (n) and α ₂ (n), corrects each BF output with 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 direction of the target area viewed from the first microphone array, the subtraction type BF 200 is, for example, as shown in Expression (9), the BF of the first microphone array. the multiplied by the amplitude spectrum correction coefficient alpha ₂ to the output _Y 1 from the (n) of the second microphone array BF output _Y 2 (n) to SS. Similarly, the subtraction type BF ₂₀₀ extracts the non-target area sound N ₂ (n) existing in the target area direction as viewed from the second microphone array according to the following equation (10).

Thereafter, the subtraction type BF 200 extracts the non-target area sound from each BF output according to the following equation (11) or (12) to extract the target area sound. The following equation (11) shows a process for extracting a target area sound with reference to 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 abnormal 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 increases as the volume levels of the background noise and the non-target area sound increase, the technique of Patent Document 3 discloses the technique of Japanese Patent Application Laid-Open No. H11-157100. Are also increased in proportion to the volume levels of the background noise and the non-target area sound.

  Specifically, in the technique 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 emphasizing the target area sound and the non-target area sound existing outside the target area direction. Calculated from the combined sound. Further, 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 level of the estimated noise and the non-target area sound.

  When the 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 is mixed with the target area sound, and it is difficult to know 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 method 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 2005-195555 A 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, Feb. 25, 2011

  However, according to the method disclosed in 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 collecting 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 collecting 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 having an area as a sound source; and (2) an input signal of any one of the microphone arrays based on a result of comparing average amplitude spectra of input signals of the respective microphone arrays. Selecting means for selecting as a mixed signal; (3) signal mixing means for mixing, as a mixed signal, the input signal of the microphone array selected by the selecting means with the target area sound extracted by the target area sound extracting means; (4) The selection means has an output means for outputting a mixed signal after mixing.

  A sound collection program according to a second aspect of the present invention provides a computer which (1) acquires beamformer outputs of the respective microphone arrays based on input signals input from a plurality of microphone arrays, and outputs the acquired beamformer outputs. A target area sound extracting means for extracting a target area sound using the target area as a sound source, and (2) comparing one of the microphone arrays based on a result of comparing average amplitude spectra of input signals of the microphone arrays. Selecting means for selecting an input signal as a mixed signal; and (3) signal mixing for mixing an input signal of the microphone array selected by the selecting means with a target area sound extracted by the target area sound extracting means as a mixed signal. And (4) functioning as output means for outputting a mixed signal mixed by the selection means. And wherein the Rukoto.

  According to a third aspect of the present invention, there is provided a sound collection method performed 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 selects one of the microphone array input signals as a mixed signal based on a result of comparing the average amplitude spectra of the input signals of the microphone arrays, and (4) the signal mixing means The input signal of the microphone array selected by the selection means is added to the target area sound extracted by the target area sound extraction means as a mixed signal. Mixed Te, (5) and the output means, and outputs the mixed signal after said selecting means are combined.

  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. 13 is a block diagram illustrating a configuration related to a conventional subtraction type BF when the number of microphones is two. FIG. 10 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 pickup device 100 performs a target area sound pickup process of picking up a target area sound from a sound source in the target area using two microphone arrays MA (MA1, 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, and 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 signal selection unit 108. , A signal mixing unit 109, and a signal output unit 110.

  Detailed processing of each functional block configuring 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-described 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 into 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 the 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 has been 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. First, the delay correction unit 104 acquires 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 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 5 according to the equation (9) or (10), and removes the non-target area sound existing in the target area direction. Extract. Further, the target area sound is extracted by subjecting the extracted noise to SS from the output of each BF according to (10) or (11).

The mixed signal selection unit 108 selects a signal having the smallest average amplitude spectrum among the input signals of the microphones constituting each microphone array as a mixed signal. When performing area sound collection using the 2-channel microphone arrays MA1 and MA2, the mixed signal selection unit 108 selects the mixed signal X _MIX (n) by processing according to the following equations (13) and (14). Can be.

In addition, in a sentence portion (text portion) other than the following mathematical formulas, for convenience of usable characters (symbols), a symbol in which “X” in mathematical formulas such as formula (13) is overlined (overline; bar) Is rewritten as “ΔX”. For example, the following expression (13) is expressed as “＾ X _ij (n) = min (＾ X ₁₁ (n), ＾ X ₁₂ (n), ＾ X ₂₁ (n), ＾ X ₂₂ (n))”. Rewritten and described.

Here, ＾ X ₁₁ (n) and ＾ X ₁₂ (n) are average amplitude spectra of the input signals (X ₁₁ (n), X ₁₂ (n)) of the microphones M1 and M2 constituting the microphone array MA1, respectively. is there. Further, ＾ X ₂₁ (n) and ＾ X ₂₂ (n) are the average amplitude spectra of the input signals (X ₂₁ (n), X ₂₂ (n)) of the microphones M1 and M2 constituting the microphone array MA2, respectively. . Further, i is the number (identifier) of the microphone array, and j is the number (identifier) of the microphones constituting the microphone array. Here, the number of the microphone array MA1 is represented by 1 and the number of the microphone array MA2 is represented by 2. Here, the number of the microphone M1 is represented by 1 and the number of the microphone M2 is represented by 2.

When X ₁₁ (n) or X ₁₂ (n) is selected as the mixed signal in the mixed signal selection unit 108, the mixed signal is output from the microphones constituting the microphone array as in the following equation (15). The average of the input signals may be used. The average amplitude spectrum may be band-limited by setting upper and lower limits of the frequency instead of the entire band.

The signal mixing unit 109 mixes the input signal selected by the mixed signal selection unit 108 with the target area sound extracted by the target area sound extraction unit 107. For example, when the signal mixing unit 109 performs area sound collection based on the microphone array MA1 according to Expression (11), the final output W ₁ (n) is mixed according to Expression (16) below. Here, μ is a parameter for adjusting the magnitude of the signal to be mixed.

  When restoring the phase to the mixed output signal in the signal mixing unit 109, the phase information includes, as the phase information, the averaging of the input signals of the microphones constituting the microphone array with reference to the target area sound extraction unit 107, or any one of them. The input signal of one microphone may be used. The signal mixing section 109 may use the phase of the input signal selected as the mixed signal.

For example, in the signal mixing section 109, when the target area sound is extracted by Expression (11), the microphone array MA1 is used as a reference. Therefore, signal mixing section 109 uses the average of input signals X ₁₁ (n) and X ₁₂ (n) or the phase information of either X ₁₁ (n) or X ₁₂ (n) to output signals. The phase may be restored.

  Further, in the signal mixing section 109, the signal mixing process may be performed 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, the signal mixing unit 109 can separate information used for phase restoration into 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 signals of the microphones constituting the microphone array used as the reference in the target area sound extracting unit 107, or any of the microphones constituting the reference microphone array. One input signal may be applied, and the phase of the input signal selected as the mixed signal 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, the mixed signal selection unit 108 compares, as a mixed signal, the average amplitude spectra of the input signals of all microphones used for area sound pickup, and selects a signal having the smallest average amplitude spectrum. 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, a non-target area sound exists near a target area by selecting a signal having the smallest average amplitude spectrum among input signals of each microphone array as a mixed signal. Also in this case, 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.

  FIG. 2 is an explanatory diagram (image diagram) showing the relationship between the position of each microphone array and the position of each sound source (the position of the sound source of the target area sound and the non-target area sound).

  In the first embodiment, assuming that the distance from each microphone array to the center of the sound pickup area is equal, the target area sound is input to all the microphones constituting each microphone array at the same volume (FIG. 2 ( a)). 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 (see FIG. 2 (b)). That is, the input signal of the microphone located farthest from the non-target area sound includes the smallest non-target area sound. Since the target area sound is included in all the microphones at the same volume, the input signal having the smallest average amplitude spectrum among all the microphones has the highest SN ratio. Therefore, in the first embodiment, even when a non-target area sound exists near the target area, the effect of suppressing the mixing of the mixed non-target area sound and improving the distortion of the target area sound is obtained. 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 11 is added. Further, the sound collecting device 100A of the second embodiment is different from the first embodiment in that the target area sound extraction unit 107 is replaced with the target area sound extraction unit 107A. Furthermore, the second embodiment is different from the first embodiment in that the processing result of the mixed signal selection unit 108 is supplied to the 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 the Second Embodiment Next, the operation (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 microphone array selection unit 111 selects a microphone array (MA1 or MA2) as a reference in the target area sound extraction unit 107 based on the mixed signal selected by the mixed signal selection unit 108.

For example, when the microphone array selection unit 111 selects the input signal X ₁₁ (n) of the microphones constituting the microphone array MA1 as the mixed signal, the microphone array MA1 has been selected as the reference microphone array. In this case, the target area sound extraction unit 107A extracts the target area sound according to equation (11).

(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 uses the microphone array (the microphone array selected by the microphone array selection unit 111) serving as the source of the mixed signal selected by the mixed signal selection unit 108 as a reference. Performs area sound extraction processing. 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, so that the distortion of the target area sound is further improved. The effect described above can be achieved.

(C) Other Embodiments The present invention is not limited to the above embodiments, but 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 number of microphones used for sound collection is three or more. 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 calculation unit, 107: target area sound extraction unit, 108: mixed signal selection unit, 109: signal mixing unit, 110: signal output unit.

According to a first aspect of the present invention, there is provided a sound collecting 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 having an area as a sound source; and (2) an input signal of any one of the microphone arrays based on a result of comparing average amplitude spectra of input signals of the respective microphone arrays. Selecting means for selecting as a mixed signal; (3) signal mixing means for mixing, as a mixed signal, the input signal of the microphone array selected by the selecting means with the target area sound extracted by the target area sound extracting means; (4) possess and output means for outputting a mixed signal after said selection means are mixed, (5) the selection means, double Average amplitude spectrum in the input signal of the microphone array and selects the minimum signal as the mixing signal.

A sound collection program according to a second aspect of the present invention provides a computer which (1) acquires beamformer outputs of the respective microphone arrays based on input signals input from a plurality of microphone arrays, and outputs the acquired beamformer outputs. A target area sound extracting means for extracting a target area sound using the target area as a sound source, and (2) comparing one of the microphone arrays based on a result of comparing average amplitude spectra of input signals of the microphone arrays. Selecting means for selecting an input signal as a mixed signal; and (3) signal mixing for mixing an input signal of the microphone array selected by the selecting means with a target area sound extracted by the target area sound extracting means as a mixed signal. And (4) functioning as output means for outputting a mixed signal mixed by the selection means. , (5) the selection means, the average amplitude spectrum in the plurality of input signals of the microphone array, characterized in you to select a minimum signal as the mixing signal.

According to a third aspect of the present invention, there is provided a sound collection method performed 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 selects one of the microphone array input signals as a mixed signal based on a result of comparing the average amplitude spectra of the input signals of the microphone arrays, and (4) the signal mixing means The input signal of the microphone array selected by the selection means is added to the target area sound extracted by the target area sound extraction means as a mixed signal. Mixed Te, (5) said output means outputs the mixed signal after said selection means are mixed, (6) the selection means, the average amplitude spectrum in the plurality of input signals of the microphone array is smallest A signal is selected as the mixed signal .

Claims (6)

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 selecting an input signal of any one of the microphone arrays as a mixed signal based on a result of comparing the average amplitude spectra of the input signals of the respective microphone arrays,
Signal mixing means for mixing the input signal of the microphone array selected by the selection means with the target area sound extracted by the target area sound extraction means as a mixed signal,
An output unit for outputting a mixed signal mixed by the selection unit.   2. The sound pickup device according to claim 1, wherein the selection unit selects a signal having a minimum average amplitude spectrum among the input signals of the plurality of microphone arrays as the mixed signal. 3.   3. The sound collection 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. 4.   4. The sound pickup according to claim 3, 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. 5. 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 selecting an input signal of any one of the microphone arrays as a mixed signal based on a result of comparing the average amplitude spectra of the input signals of the respective microphone arrays,
Signal mixing means for mixing the input signal of the microphone array selected by the selection means with the target area sound extracted by the target area sound extraction means as a mixed signal,
A sound collection program characterized by causing the selection means to function as an output means for outputting a mixed signal after mixing. 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 area sounds,
The selecting means selects any one of the microphone array input signals as a mixed signal based on a result of comparing the average amplitude spectrum of the respective microphone array input signals,
The signal mixing unit mixes the input signal of the microphone array selected by the selection unit with the target area sound extracted by the target area sound extraction unit as a mixed signal,
The sound pickup method, wherein the output means outputs a mixed signal mixed by the selection means.

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