JP2024027222A - Generation device, generation method and program - Google Patents

Abstract

To provide a generation device, etc. which achieves high suppression performance even in the ear of a user who is far from an actual error microphone.SOLUTION: A generation device includes: a sound pressure estimation part which generates a cancel signal so that a point in which a suppression amount of noise becomes maximum is located to a user than installation positions of a plurality of error microphones which are arranged on a spherical face close to the head of a user, estimates a sound collection signal collected when a virtual microphone is installed in a place closer to an observation point than the plurality of error microphones by using a spherical harmonic function expansion coefficient and obtains an estimation sound collection signal x(v); and a suppression signal generation part for generating the cancel signal for suppressing noise in the installation position of the virtual microphone by using a sound collection signal x(r) which collects noise of a suppression object and the estimation sound collection signal x(v). The virtual microphone is the microphone which is not actually installed but is virtually installed.SELECTED DRAWING: Figure 4

Description

Application for application of Article 30, Paragraph 2 of the Patent Act (1) Publication date: Publication of the proceedings February 23, 2022, Date of invention announcement: March 9, 2022 (Holding date: March 9, 2022 - March 11, 2022) Publication Acoustical Society of Japan 2022 Spring Research Presentation (Proceedings) https://acoustics. jp/annualmeeting/program/

The present invention relates to an active noise control (ANC) technique that suppresses external noise at a specific location.

Non-patent document 1 is known as a conventional active noise suppression technique. Active noise suppression commonly uses reference microphones, error microphones, and canceling speakers. FIG. 1 shows an example of the configuration of a conventional noise suppression device. The reference microphone 91 picks up the noise emitted by the noise source. The cancellation speaker 92 reproduces the cancellation signal generated by the suppression signal generation device 90 and emits a cancellation sound that cancels out the noise. Further, the error microphone 93 picks up the unmated noise and feeds it back. The suppression signal generation device 90 uses the sound signal picked up by the reference microphone 91 and the sound picked up signal from the error microphone 93 to actively control and generate a cancellation signal so that less noise is left unerased. At the installation position of the error microphone 93, the cancellation speaker 92 emits a cancellation sound so that the amount of noise left unerased is reduced, so that the cancellation sound suppresses noise most efficiently at the installation position of the error microphone 93. Therefore, the error microphone 93 is installed near the user's ear.

However, in actual use, it may not be possible to install the error microphone 93 close to the user's ear, and if the distance between the installation position of the error microphone 93 and the user's ear increases, the installation position of the error microphone 93 may change as described above. The noise is suppressed most efficiently in the user's ears, but the noise remains unmated near the user's ears, the suppression performance deteriorates, and the user may not be able to fully benefit from the noise suppression. For example, if the distance from the noise source to the user's ear is 100 mm, and the error microphone 93 is installed near the user's ear (0 mm), the suppression performance is -∞dB, and the error microphone 93 is placed at the midpoint between the noise source and the user's ear. It was confirmed through simulation that the suppression performance was -7.38dB when installed at FIG. 2 is a diagram for explaining the difference between the suppressable region (sweet spot) S ₁ of the prior art and the desired sweet spot S ₂ .

In Non-Patent Document 2, as shown in FIG. 3, a plurality of error microphones 93 are installed around the head so as to cover the head, and the sound pressure inside the ear is estimated from the signal obtained by the error microphones 93. Predict sound pressure.

Kajikawa, “Recent topics and applications of active noise control,” Research Report Music Information Science (MUS), vol. 2015-MUS-107, no. 3, pp. 1-6, May 2015. Nobuo et al., “Study of ANC that predicts the sound pressure at the ear using multiple microphones near the head assuming a hard sphere,” Proceedings of the Acoustical Society of Japan (Spring), pp. 433-434, March 2022.

However, there is a problem in that the conditions for installing a plurality of error microphones are not realistic when considering use at seats in public institutions.
The present invention estimates the sound signal obtained when sound is collected near the user's ear from the sound signal collected by an error microphone installed based on installation conditions suitable for use in seats in public institutions, etc., In order to actively control the cancellation signal, the estimated sound signal is used instead of the sound signal collected at the actual location of the error microphone, allowing the user to move away from the actual error microphone. The purpose of the present invention is to provide a generation device, a generation method, and a program therefor that achieve high suppression performance even when close to the user's ear.

In order to solve the above problems, according to one aspect of the present invention, a generation device generates a cancellation signal used for active noise control. The generation device generates a cancellation signal such that the point where the amount of noise suppression is maximum is located closer to the user than the installation position of the plurality of error microphones, and the plurality of error microphones are located close to the user's head. The virtual microphone is placed on a spherical surface and uses the spherical harmonic expansion coefficients to estimate the sound signal that would be picked up if the virtual microphone was placed closer to the observation point than multiple error microphones. Suppress the noise at the installation position of the virtual microphone using a sound pressure estimator that obtains x(v), a picked-up signal x(r) that collects the noise to be suppressed, and an estimated picked-up signal x(v). The virtual microphone is a microphone that is not actually installed but is installed virtually.

According to the present invention, when an error microphone cannot be placed near the user's ear, it is possible to achieve high suppression performance without installing multiple error microphones around the user's head so as to cover the user's head. be effective.

A diagram for explaining conventional active noise control. FIG. 3 is a diagram for explaining a suppressable region in the conventional technology. FIG. 3 is a diagram for explaining the installation position of an error microphone according to the prior art. FIG. 1 is a functional block diagram of the noise suppression system according to the first embodiment. The figure which shows the example of the process flow of the noise suppression system based on 1st embodiment. FIG. 3 is a diagram for explaining estimation method 1. FIG. 3 is a diagram for explaining estimation method 1. FIG. 3 is a diagram for explaining estimation method 2. FIG. 7 is a diagram for explaining estimation method 3. FIG. 4 is a diagram for explaining estimation method 4. FIG. 4 is a diagram for explaining estimation method 4. FIG. 7 is a diagram for explaining estimation method 5. FIG. 7 is a diagram for explaining estimation method 7. The figure which shows the example of a structure of the computer to which this method is applied.

Embodiments of the present invention will be described below. In the drawings used in the following explanation, components having the same functions and steps that perform the same processing are denoted by the same reference numerals, and redundant explanation will be omitted. In the following description, processing performed for each element of a vector or matrix is applied to all elements of that vector or matrix, unless otherwise specified.

<Points of the first embodiment>
In this embodiment, the observed sound pressure at the user's ear is estimated from the sound signals picked up by a plurality of error microphones installed on a spherical surface close to the user's head. For example, the pickup signal of a virtual error microphone placed near the ear is estimated from the pickup signals of multiple actual error microphones, and in ANC, the pickup signal of the virtual error microphone is compared to that of the conventional error microphone. Used as a sound signal. With this configuration, it is possible to change the sweet spot position from the installation position of the error microphone to the virtual error microphone position, and produce a sound that cancels the unerased sound near the ear.

Various methods can be considered for estimating a sound signal picked up by a virtual error microphone. For example, the sound pressure at the ear is estimated using a spherical harmonic function from a plurality of actual error microphones placed on a spherical surface close to the user's head.
Note that the plurality of error microphones are arranged so that the center of the spherical surface on which the plurality of error microphones are arranged is located outside the user's head.

<First embodiment>
FIG. 4 shows a functional block diagram of the noise suppression system according to the first embodiment, and FIG. 5 shows its processing flow.

The noise suppression system includes a reference microphone 91, a cancellation speaker 92, a microphone array including L error microphones 93-i, a suppression signal generation section 110, and a sound pressure estimation section 120. Let i=1,2,…,L. The device including the suppression signal generation section 110 and the sound pressure estimation section 120 is also referred to as a suppression signal generation device.

The suppression signal generation device inputs the sound signal x(r) of the reference microphone 91 and the sound signal x(e) of the L error microphones 93-i, and determines the point where the amount of noise suppression is maximum. , a cancellation signal (hereinafter also referred to as a "suppression signal") y is generated and output to the cancellation speaker 92 so as to be located closer to the user than the installation position of the error microphone 93-i.

The suppression signal generation device is, for example, a special computer configured by loading a special program into a known or dedicated computer having a central processing unit (CPU), a main memory (RAM), etc. It is a great device. The suppression signal generation device executes each process under the control of, for example, a central processing unit. The data input to the suppression signal generation device and the data obtained through each process are stored, for example, in the main memory, and the data stored in the main memory is read out to the central processing unit as necessary. Used for other processing. Each processing unit of the suppression signal generation device may be configured at least in part by hardware such as an integrated circuit. Each storage unit included in the suppression signal generation device can be configured by, for example, a main storage device such as a RAM (Random Access Memory), or middleware such as a relational database or a key-value store. However, each storage unit does not necessarily have to be provided with a suppression signal generation device therein, and may be configured with an auxiliary storage device composed of a hard disk, an optical disk, or a semiconductor memory element such as a flash memory, and the suppression signal generation device It may be configured to be provided outside the generation device.

Each part will be explained below.

<Reference microphone 91>
The reference microphone 91 picks up the sound to be suppressed (S91) and outputs a picked-up signal x(r). The sound to be suppressed that is picked up by the reference microphone 91 will be referred to as "noise" hereinafter.

<Cancel speaker 92>
The cancellation speaker 92 inputs the cancellation signal y and reproduces the cancellation signal y (S92). When the reproduced sound reproduced from the cancellation speaker 92 and the noise to be suppressed are completely opposite in phase, the reproduced sound and the noise to be suppressed overlap, that is, the sound waves are superimposed on each other, and the waves cancel each other out. Noise is suppressed.

<Error microphone 93-i>
The error microphone 93-i picks up sounds that are not suppressed by the reproduced sound played from the canceling speaker 92, including unmated noise (S93), and outputs a picked-up sound signal x(e). The error microphone 93-i is placed closer to the noise source than the observation point (for example, near the user's ear). The positions of the L error microphones 93-i are related to the sound pressure estimation method, so they will be explained together with the sound pressure estimation section 120, which will be described later.

<Sound pressure estimation unit 120>
The sound pressure estimation unit 120 inputs the output signals (sound collection signals) x(e) of the L error microphones 93-i, and places the microphone 130 at a position closer to the observation point than the L error microphones 93-i. An estimated sound pickup signal x(v), which is a signal estimated to be picked up when installed, is calculated and output. That is, the sound pressure estimating unit 120 estimates a sound signal obtained when the sound that is not suppressed by the reproduced sound played from the canceling speaker 92 is collected at the installation position of the microphone 130 (S120), and estimates The collected sound signal is output as the estimated sound collected signal x(v). Below, seven methods for estimating the estimated sound pickup signal x(v) will be illustrated. Here, the microphone 130 is not actually installed but is installed virtually, and is hereinafter referred to as a virtual microphone 130.

(Estimation method 1)
In this estimation method, a sound signal picked up by a virtual error microphone is estimated from sound signals picked up by a plurality of error microphones placed on a spherical surface close to the user's head, using spherical harmonic expansion coefficients. The conditions of estimation method 1 will be explained below.

(Condition 1)
The plurality of error microphones are arranged on the spherical surface of a virtual sphere (hereinafter also referred to as virtual sphere B) close to the user's head. A virtual sphere is not a physically existing sphere. Further, the plurality of error microphones are arranged so that the center of the virtual sphere B is located outside the user's head.

(Condition 2)
In order to use the spherical harmonic function expansion coefficient, the number L of error microphones and the expansion order N of the spherical harmonic function satisfy (N+1) ² ≦L.

(Condition 3)
The position of the virtual microphone is the position of the user's ear, and is on the surface or outside of the virtual sphere. Note that estimating the sound pressure at a point located on the surface or outside of the virtual sphere is called estimation based on sound pressure extrapolation.

6 and 7 are diagrams for explaining the positional relationship of the plurality of error microphones 93-i on the spherical surface of the virtual sphere B close to the user's head.

In this estimation method, an origin O' is placed at a predetermined distance from the position of the listener's ear, and a plurality of error microphones are placed at equidistant positions (positions with the same radius) from the origin O'. That is, an error microphone is placed on the surface of virtual sphere B. An error microphone is placed on the spherical surface of virtual sphere B with radius a, and the sound pressure on the spherical surface with radius r is estimated. For example, the distance from the center (origin O') to the error microphone is set to a = 7.5 cm, and four error microphones are placed at the vertices of a triangular pyramid inscribed in a virtual sphere with radius a. At this time, the four error microphones form a tetra-shaped microphone array. For example, the distance from the center (origin O') to the observation point (position of virtual microphone 130) is estimated as r=8 cm.

By using spherical harmonic expansion, it is possible to estimate the observed sound pressure on any spherical surface from the observed sound pressure on a certain spherical surface.

The sound pressure p(a,Ω _o ) observed at a certain angle Ω _o =(θ _o ,φ _o ) on the surface of a sphere of radius a is, if the noise comes only from outside the sphere,

It can be expressed as Here, k is a wave number, β _n,m is a sound field coefficient specific to the noise field, j _n is an n-th order spherical Bessel function, and Y ^m _n (·) is a spherical harmonic function.

The sound field coefficient β _n,m is calculated using spherical harmonic expansion.

It can be calculated as

Therefore, the sound pressure p(r,Ω) at an arbitrary angle Ω of radius r that does not include the noise source is

It can be estimated by

Therefore, the sound pressure estimation unit 120 calculates the observed sound pressure values [p(a, Ω ₁ ), p(a, Ω ₂ ),…,p(a,Ω _L )], find the sound field coefficient β _n,m using equation (2), and use the sound field coefficient β _n,m to calculate the virtual microphone using equation (3). Estimate the sound pressure p(r,Ω) at . In addition, in order to reduce the amount of calculation, the sound field coefficients β _n,m corresponding to various observed values of sound pressure are calculated in advance, and the combinations of the observed values of sound pressure and the sound field coefficients β _n,m are illustrated. When estimating the sound pressure, the observed value of the sound pressure obtained from the output signals (collected sound signals) x(e) of the L error microphones 93-i is stored in the storage section [p(a ,Ω ₁ ),p(a,Ω ₂ ),…,p(a,Ω _L )], the sound field coefficient β _n,m corresponding to You can also use it as Furthermore, if the position (r, Ω) of the virtual microphone is determined in advance, the sound pressure p(r, Ω) at the virtual microphone corresponding to various observed values of sound pressure is determined in advance, and the sound pressure The combination of the observed value x[e]=[p(a,Ω ₁ ),p(a,Ω ₂ ),…,p(a,Ω _L )] and the sound pressure p(r,Ω) at the virtual microphone is The observed value of the sound pressure [p( a,Ω ₁ ),p(a,Ω ₂ ),…,p(a,Ω _L )] at the virtual microphone, and calculate the sound pressure p(r,Ω) at the virtual microphone without performing calculation processing. It may also be configured to estimate the sound pressure.

Note that in the case of L=4, N is limited to the first order in order to satisfy (N+1) ² ≦L.
On the other hand, when expressing a sound field using spherical harmonic expansion, it is known that frequencies and radii in the range kr<N can be expressed with high accuracy. Here, r is the radius (distance to the virtual microphone), k = 2πf/c (c is the speed of sound = 340 m/s, f is the frequency), so for example, if N = 1, the radius (distance to the virtual microphone) It can be seen that within a range of 7.5 cm, frequencies up to 721 Hz can be expressed and predicted. Noise is concentrated in the low frequency range, so if frequencies up to 721Hz can be predicted, it would be useful.

(Estimation method 2)
First, the conditions for estimation method 2 will be explained.
Estimation method 2 also needs to satisfy conditions 1 and 2 described in estimation method 1. In estimation method 2, instead of condition 3, it is necessary to satisfy condition 4 below.

(Condition 4)
The position of the virtual microphone is the position of the user's ear and the surface of the virtual sphere.

Therefore, in this estimation method, as in estimation method 1, the origin O' is placed at a predetermined distance from the listener's ear position, and multiple Place the error microphone. That is, an error microphone is placed on the surface of the virtual sphere. Furthermore, a plurality of error microphones are arranged so that the distance from the origin O' to the user's ear is the same as the radius of the virtual sphere B (see FIG. 8). At this time, when substituting equation (2) into equation (3), r and a become the same, so the division of the spherical Bessel function j _n (・) disappears from the prediction equation, and the estimated sound pressure is can be found.

The sound pressure estimation unit 120 calculates the observed sound pressure values [p(a,Ω ₁ ), p(a,Ω ₂ ),…,p(a,Ω _L )] to find the sound field coefficient P _n,m using equation (4), and using the sound field coefficient P _n,m to calculate the sound pressure at the virtual microphone using equation (5). Estimate p(r,Ω). Note that the configuration for reducing the amount of calculation described in estimation method 1 may be adopted by using the sound field coefficient P n _,m instead of the sound field coefficient β n, _m .

(Estimation method 3)
First, the conditions for estimation method 3 will be explained.
Estimation method 3 also needs to satisfy condition 2 explained in estimation method 1. In estimation method 3, instead of conditions 1 and 3, the following conditions 5 and 6 need to be satisfied.

(Condition 5)
The plurality of error microphones are arranged at equal intervals on the spherical surface of the virtual sphere B close to the user's head.

(Condition 6)
The position of the virtual microphone is closer to the center of the virtual sphere than the position of the user's ear, and inside the virtual sphere. Note that estimating the sound pressure at a point located inside the virtual sphere is called estimation based on sound pressure interpolation.

Therefore, in this estimation method, as in estimation method 1, the origin is placed at a predetermined distance from the listener's ear position, and multiple error microphones are placed at equal intervals from the origin position (positions with the same radius). Place. Furthermore, a plurality of error microphones are arranged at equal intervals on the spherical surface of virtual sphere B. For example, the points of contact between a regular polyhedron and its circumscribed sphere (outer radius a) (each vertex of the regular polyhedron), and the points of contact between a regular polyhedron and its inscribed sphere (inscribed radius a) (the center of each face of the regular polyhedron) By placing the error microphones at equal intervals, the error microphones can be placed at equal intervals. For example, error microphones are placed at each vertex of a regular tetrahedron inscribed in the virtual sphere.

If the number of error microphones is L and the L error microphones are arranged at equal intervals on the virtual sphere, the maximum order N of n becomes (N+1) ² ≦L, and the integral becomes the sum. The sound field coefficient β _n,m is calculated using the following formula.

When predicting the sound pressure at the ear position using this sound field coefficient β _n,m , instead of using the distance r from the origin of the virtual sphere to the ear position, we calculate the distance r _d < r to estimate sound pressure.

FIG. 9 shows the relationship between the user's ear position and the positions of the virtual sphere, error microphone, and virtual microphone. In estimation methods 1 and 2, the user's ear position and the virtual microphone position match, but in this estimation method, the user's ear position and the virtual microphone position are different.

The sound pressure estimation unit 120 calculates the observed sound pressure values [p(a,Ω ₁ ), p(a,Ω ₂ ),…,p(a,Ω _L )], the sound field coefficient β _n,m is determined by equation (6), and the sound field coefficient β n,m is used to calculate the sound field coefficient β _n,m by equation (7). Estimate the pressure p(r _d ,Ω). Note that the configuration for reducing the amount of calculation described in estimation method 1 may be adopted. However, the sound pressure p(r _d , Ω) is estimated instead of the sound pressure p(r, Ω).

For example, by combining this estimation method and estimation method 2, condition 6 may be changed to condition 6A as follows.

(Condition 6A)
The position of the user's ear is the surface of the virtual sphere, and the position of the virtual microphone is closer to the center of the virtual sphere than the position of the user's ear and is inside the virtual sphere.

The virtual microphone is placed inside the virtual sphere by (rr _d ) from the user's ear position. When the radius a of the virtual sphere is 7.5 cm, by setting 0<rr _d ≦6 (in terms of the ratio to the radius a of the virtual sphere, 0<rr _d ≦0.8a), the result is equivalent to or better than estimation method 1. The prediction accuracy was particularly high when rr _d =3 (r _d =0.4a in terms of the ratio to the radius a of the virtual sphere).

(Estimation method 4)
First, the conditions of estimation method 4 will be explained.
Estimation method 4 also needs to satisfy conditions 1 and 3 explained in estimation method 1. In estimation method 4, condition 2 does not have to be satisfied.

In this estimation method, of the microphones on the spherical surface, only the three microphones closest to the ears are used (see FIGS. 10 and 11). In this case, condition 2 is not satisfied and the sound field coefficients cannot be found analytically, so they are estimated using the least squares method or the like.

In this estimation method, error microphones are placed at non-uniform intervals on a spherical surface of radius a, and the sound pressure on the spherical surface of radius r is estimated. In this estimation method, since the spherical harmonic expansion cannot be used directly, the spherical harmonic expansion coefficients are estimated by the least squares method to obtain the sound pressure on the spherical surface of radius r.

The sound pressure estimation unit 120 calculates the observed sound pressure values [p(a,Ω ₁ ), p(a,Ω ₂ ),…,p(a,Ω _L )], the spherical harmonic expansion coefficients are estimated by the least squares method, and the sound pressure p(r,Ω) at the virtual microphone is estimated using the spherical harmonic expansion coefficients. do. Note that the configuration for reducing the amount of calculation described in estimation method 1 may be adopted by using spherical harmonic expansion coefficients instead of the sound field coefficients β _n,m .

(Estimation method 5)
First, the conditions of estimation method 5 will be explained.
Estimation method 5 also needs to satisfy condition 2 explained in estimation method 1. In estimation method 5, conditions 7 and 8 are satisfied instead of conditions 1 and 3.

(Condition 7)
The plurality of error microphones are placed on the spherical surface of the hard sphere B' close to the user's head. The hard sphere B' is an object that physically exists and reflects sound (see FIG. 12).

(Condition 8)
The position of the virtual microphone is the position of the user's ear and outside the virtual sphere.

In this estimation method, the sound pressure is estimated by assuming that the sphere is not a virtual sphere but a hard sphere that reflects sound.
By arranging multiple error microphones on the spherical surface of the rigid sphere B', the radius of the sphere can be reduced. For example, in estimation method 1, if the radius a of the virtual sphere B is 7.5 cm, the radius a' of the hard sphere B' is 3 cm or more and less than 7.5 cm (in terms of the ratio to the radius a of the virtual sphere, it is 0.4 a By setting ≦a′<1), it is possible to achieve prediction accuracy comparable to or higher than that of estimation method 1. The prediction formula in this case is as follows.

a is the radius of the hard sphere, j _n ' is the differential of j _n , h _n ⁽²⁾ is the Hankel function of the second kind, and h _n ' ⁽²⁾ is the differential of h _n ⁽²⁾ .

The sound pressure estimation unit 120 calculates the observed sound pressure values [p(a,Ω ₁ ), p(a,Ω ₂ ),…,p(a,Ω _L )], the sound field coefficient B _n,m is determined by equation (9), and the sound field coefficient B n,m is used to calculate the sound field coefficient B _n,m by equation (8). Estimate the pressure p(r,Ω). Note that the configuration for reducing the amount of calculation described in estimation method 1 may be adopted by using the sound field coefficient B n, _{m instead of the sound field coefficient β n, m} . Furthermore, this estimation method and estimation method 3 may be combined.

(Estimation method 6)
First, the conditions of estimation method 6 will be explained.
Estimation method 6 can be combined with any of estimation methods 1 to 5, and estimation method 6 needs to satisfy the conditions of any of estimation methods 1 to 5. Furthermore, in estimation method 6, the following condition 9 is satisfied.

(Condition 9)
Use a directional microphone instead of an omnidirectional microphone as an error microphone.

By using a directional microphone, the radius of virtual sphere B can be made as small as possible, and sound field coefficients can be directly determined and extrapolated according to Ambisonics.

The sound pressure estimation unit 120 calculates the observed sound pressure values x[e]=[p(a,Ω ₁ ),p(a,Ω ₂ ),...,p( a,Ω _L )] to find the sound field coefficient β _n,m according to ambisonics, and use the sound field coefficient β _n,m to estimate the sound pressure p(r, Ω) at the virtual microphone. Note that the configuration for reducing the amount of calculation described in estimation methods 1 to 5 may be adopted.

(Estimation method 7)
First, the conditions of estimation method 7 will be explained.
It is necessary to satisfy the conditions of estimation method 7 and estimation method 5. Furthermore, estimation method 7 satisfies condition 10 below.

(Condition 10)
At least one or more speakers are embedded in the rigid sphere B'.

By using one or more speakers 94-m, ANC and local playback based on ear sound pressure prediction can be achieved with a single device (see FIG. 13). m=1,2,...,M, where M is the number of speakers embedded in the hard sphere B'.

By installing the speaker inside the hard sphere B', we can satisfy the requirements of Kirchhoff's integral equation (we do not want the sound source to be inside the radius to be extrapolated) when deriving the sound pressure prediction formula, and the sound pressure This can be expected to have the effect of increasing the extrapolation accuracy.

The sound pressure estimation unit 120 calculates the observed sound pressure values [p(a,Ω ₁ ), p(a,Ω ₂ ),…,p(a,Ω _L )], the sound field coefficient β _n,m is determined by equation (9), and the sound field coefficient β n,m is used to calculate the sound field coefficient β _n,m by equation (8). Estimate the pressure p(r,Ω). Note that the configuration for reducing the amount of calculation described in estimation method 1 may be adopted.

<Suppression signal generation unit 110>
The suppression signal generation unit 110 receives the collected sound signal x(r) and the estimated collected sound signal x(v) as input, and generates a cancellation signal y for suppressing noise at the installation position of the virtual microphone 130 (S110). ,Output.

Conventional techniques can be used as a method for generating the cancellation signal. For example, the method disclosed in Non-Patent Document 1 can be used. In this embodiment, a feedforward ANC is realized using a picked-up sound signal x(r), an estimated picked-up signal x(v), and a cancellation signal y. In addition to estimating the sound pickup signal that would be obtained when the virtual microphone 130 detects the interference sound between the noise from the noise source and the reproduced sound of the cancellation signal y, the noise from the noise source is detected by the reference microphone 91, A cancellation signal y is generated by inputting it to a noise control filter realized by an adaptive digital filter, and is reproduced by the cancellation speaker 92. It is assumed that the reproduced sound of the cancellation signal y propagates through a secondary path that is a series of transmission systems from the cancellation speaker 92 to the virtual microphone 130. Then, the coefficients of the noise control filter are updated using an adaptive algorithm so that the input to the virtual microphone 130 is minimized. As a method for updating the coefficients of the noise control filter, a conventional updating method can be used, so a description thereof will be omitted. In feedforward ANC, a secondary path model that estimates the secondary path is used to compensate for the influence of the secondary path in an adaptive algorithm.

<Effect>
With the above configuration, when an error microphone cannot be placed near the user's ear, high suppression performance can be achieved without installing a plurality of error microphones around the user's head so as to cover the user's head.

<Other variations>
The present invention is not limited to the above-described embodiments and modifications. For example, the various processes described above may not only be executed in chronological order as described, but may also be executed in parallel or individually depending on the processing capacity of the device executing the process or as needed. Other changes may be made as appropriate without departing from the spirit of the present invention.

<Program and recording medium>
The various processes described above are performed by causing the recording unit 2020 of the computer 2000 shown in FIG. This can be done by letting

A program describing the contents of this process can be recorded on a computer-readable recording medium. The computer-readable recording medium may be of any type, such as a magnetic recording device, an optical disk, a magneto-optical recording medium, or a semiconductor memory.

Further, this program is distributed by, for example, selling, transferring, lending, etc. a portable recording medium such as a DVD or CD-ROM on which the program is recorded. Furthermore, this program may be distributed by storing the program in the storage device of the server computer and transferring the program from the server computer to another computer via a network.

A computer that executes such a program, for example, first stores a program recorded on a portable recording medium or a program transferred from a server computer in its own storage device. When executing a process, this computer reads a program stored in its own recording medium and executes a process according to the read program. In addition, as another form of execution of this program, the computer may directly read the program from a portable recording medium and execute processing according to the program, and furthermore, the program may be transferred to this computer from the server computer. The process may be executed in accordance with the received program each time. In addition, the above-mentioned processing is executed by a so-called ASP (Application Service Provider) type service, which does not transfer programs from the server computer to this computer, but only realizes processing functions by issuing execution instructions and obtaining results. You can also use it as Note that the program in this embodiment includes information that is used for processing by an electronic computer and that is similar to a program (data that is not a direct command to the computer but has a property that defines the processing of the computer, etc.).

Further, in this embodiment, the present apparatus is configured by executing a predetermined program on a computer, but at least a part of these processing contents may be realized by hardware.

Claims (8)

A generation device that generates a cancellation signal used for active noise control,
A cancellation signal is generated so that the point where the amount of noise suppression is maximum is located closer to the user than the installation positions of the plurality of error microphones,
The plurality of error microphones are arranged on a spherical surface close to the user's head,
Using the spherical harmonic function expansion coefficients, estimate the sound signal that would be picked up when the virtual microphone is installed at a position closer to the observation point than the plurality of error microphones, and calculate the estimated sound pickup signal x(v). a sound pressure estimation unit that obtains the sound pressure;
a suppression signal that generates a cancellation signal for suppressing noise at the installation position of the virtual microphone, using a picked-up sound signal x(r) that picks up the noise to be suppressed and the estimated picked-up signal x(v); and a generating section;
The virtual microphone is a microphone that is not actually installed but is installed virtually.
generator. The generating device according to claim 1,
The plurality of error microphones are arranged on a spherical surface of a virtual sphere close to the user's head, and arranged such that the center of the virtual sphere is located outside the user's head. ,
The number L of the error microphones and the expansion order N of the spherical harmonic function satisfy (N+1) ² ≦L,
The position of the virtual microphone is the position of the user's ear, and the surface or outside of the virtual sphere.
generator. The generating device according to claim 2,
The position of the virtual microphone is the position of the user's ear and the surface of the virtual sphere,
generator. The generating device according to claim 1,
The plurality of error microphones are arranged at equal intervals on the spherical surface of a virtual sphere close to the user's head, and the center of the virtual sphere is located outside the user's head. placed in
The number L of the error microphones and the expansion order N of the spherical harmonic function satisfy (N+1) ² ≦L,
The position of the virtual microphone is inside the virtual sphere,
generator. The generating device according to claim 1,
The number of error microphones is 3,
The three error microphones are arranged on the spherical surface of a virtual sphere close to the user's head, and arranged so that the center of the virtual sphere is located outside the user's head. is,
The position of the virtual microphone is the position of the user's ear, and the surface or outside of the virtual sphere,
The sound pressure estimator estimates spherical harmonic expansion coefficients by a least squares method, and uses the estimated spherical harmonic expansion coefficients to obtain the estimated sound pickup signal x(v).
generator. The generating device according to claim 1,
The plurality of error microphones are arranged on the spherical surface of a hard sphere that reflects sound close to the user's head,
The number L of the error microphones and the expansion order N of the spherical harmonic function satisfy (N+1) ² ≦L,
The position of the virtual microphone is the position of the user's ear and outside the hard sphere.
generator. A generation method for generating a cancellation signal used for active noise control, comprising:
A cancellation signal is generated so that the point where the amount of noise suppression is maximum is located closer to the user than the installation positions of the plurality of error microphones,
The plurality of error microphones are arranged on a spherical surface close to the user's head,
Using the spherical harmonic function expansion coefficients, estimate the sound signal that would be picked up when the virtual microphone is installed at a position closer to the observation point than the plurality of error microphones, and calculate the estimated sound pickup signal x(v). a step of estimating the sound pressure to obtain;
a suppression signal that generates a cancellation signal for suppressing noise at the installation position of the virtual microphone, using a picked-up sound signal x(r) that picks up the noise to be suppressed and the estimated picked-up signal x(v); a generating step;
The virtual microphone is a microphone that is not actually installed but is installed virtually.
Generation method. A program for causing a computer to function as the generating device according to any one of claims 1 to 6.

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