JP7709142B2 - GENERATION DEVICE, GENERATION METHOD, AND PROGRAM - Google Patents

Description

Article 30, paragraph 2 of the Patent Act applies (1) Publication date: Proceedings publication: February 23, 2022; date of announcement of invention: March 9, 2022 (date of holding: March 9-11, 2022) Publication: Acoustical Society of Japan 2022 Spring Research Presentation (Proceedings) https://acoustics.jp/annualmeeting/program/

The present invention relates to active noise control (ANC) technology that suppresses external noise at a specific position.

Non-Patent Document 1 is known as a conventional active noise suppression technology. In active noise suppression, a reference microphone, an error microphone, and a cancellation speaker are generally used. FIG. 1 shows an example of the configuration of a conventional noise suppression device. The reference microphone 91 picks up noise emitted by a 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 the noise. Furthermore, the error microphone 93 picks up the remaining noise and feeds it back. The suppression signal generation device 90 actively controls and generates a cancellation signal using the collected signal of the reference microphone 91 and the collected signal of the error microphone 93 so that the remaining noise is reduced. Since the cancellation speaker 92 emits the cancellation sound so that the remaining noise is reduced at the installation position of the error microphone 93, the cancellation sound most efficiently suppresses noise at the installation position of the error microphone 93. For this reason, the error microphone 93 is installed close to the user's ear.

However, in actual use, there are cases where the error microphone 93 cannot be installed close to the user's ear, and if the distance between the installation position of the error microphone 93 and the user's ear is large, as described above, the noise is most efficiently suppressed at the installation position of the error microphone 93, and the noise remaining at the user's ear increases, the suppression performance decreases, and the user may not be able to fully benefit from the noise suppression. For example, it was confirmed by simulation that when the distance from the noise source to the ear is 100 mm and the error microphone 93 is installed at the user's ear (0 mm), the suppression performance is -∞ dB, and when the error microphone 93 is installed at the midpoint between the noise source and the ear, the suppression performance is -7.38 dB. Figure 2 is a diagram for explaining the difference between the suppressible region (sweet spot) S ₁ of the conventional technology and the desired sweet spot S ₂ .

In Non-Patent Document 2, as shown in FIG. 3, multiple error microphones 93 are placed around the head to cover the head, and the sound pressure at the ear is predicted by estimating the internal sound pressure from the signals obtained by the error microphones 93.

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., "A study on ANC using multiple microphones near the head assuming a hard sphere to predict sound pressure near the ears," Proceedings of the Acoustical Society of Japan (Spring), pp. 433-434, March 2022.

However, when considering use in seats in public institutions, etc., there is a problem that the conditions for installing multiple error microphones are not realistic.
The present invention aims to provide a generation device, generation method, and program thereof that estimates a picked-up signal that would be obtained if sound were picked up near a user's ear from a picked-up signal picked up by an error microphone installed based on installation conditions suitable for use in seats in public institutions, etc., and uses the estimated picked-up signal instead of the picked-up signal picked up at the actual location of the error microphone in order to actively control a cancellation signal, thereby achieving high suppression performance even near a user's ear that is far from the actual error microphone.

In order to solve the above problem, according to one aspect of the present invention, a generating device generates a cancellation signal used in active noise control. The generating device generates a cancellation signal so that the point where the amount of noise suppression is maximum is located closer to the user than the installation positions of the multiple error microphones, and the multiple error microphones are arranged on a sphere close to the user's head. The generating device includes a sound pressure estimating unit that uses spherical harmonic function expansion coefficients to estimate a pickup signal that would be picked up if a virtual microphone were installed at a position closer to the observation point than the multiple error microphones, and obtains an estimated pickup signal x(v), and a suppression signal generating unit that uses the pickup signal x(r) that picks up the noise to be suppressed and the estimated pickup signal x(v) to generate a cancellation signal for suppressing the noise at the installation position of the virtual microphone, and the virtual microphone is a microphone that is virtually installed without actually being installed.

The present invention has the advantage that, when an error microphone cannot be placed near the user's ear, high suppression performance can be achieved without placing multiple error microphones around the user's head so as to cover the head.

FIG. 1 is a diagram for explaining a conventional active noise control. FIG. 1 is a diagram for explaining a suppressible region of the conventional technique. FIG. 1 is a diagram for explaining the installation position of an error microphone according to the prior art. 1 is a functional block diagram of a noise suppression system according to a first embodiment. FIG. 2 is a diagram showing an example of a processing flow of the noise suppression system according to the first embodiment. FIG. 13 is a diagram for explaining an estimation method 1. FIG. 13 is a diagram for explaining an estimation method 1. FIG. 13 is a diagram for explaining an estimation method 2. FIG. 13 is a diagram for explaining estimation method 3. FIG. 13 is a diagram for explaining estimation method 4. FIG. 13 is a diagram for explaining estimation method 4. FIG. 13 is a diagram for explaining estimation method 5. FIG. 13 is a diagram for explaining estimation method 7. FIG. 13 is a diagram showing an example of the configuration of a computer to which the present technique is applied.

The following describes an embodiment of the present invention. In the drawings used in the following description, components having the same functions and steps performing the same processing are denoted with the same reference numerals, and duplicate explanations will be omitted. In the following description, processing performed on each element of a vector or matrix is assumed to apply to all elements of that vector or matrix, unless otherwise specified.

<Key Points of the First Embodiment>
In this embodiment, the observed sound pressure at the ear is estimated from the pickup signals of multiple error microphones installed on a spherical surface close to the user's head. For example, the pickup signals of a virtual error microphone placed at the ear are estimated from the pickup signals of the actual multiple error microphones, and the pickup signals of the virtual error microphone are used as the pickup signals of the conventional error microphone in the ANC. With this configuration, the position of the sweet spot can be changed from the installation position of the error microphone to the position of the virtual error microphone, and a sound that cancels the residual sound at the ear can be produced.

There are various methods for estimating the sound pickup signal of the virtual error microphone. For example, the sound pressure at the ear is estimated using a spherical harmonic function from multiple actual error microphones placed on a sphere close to the user's head.
The error microphones are arranged so that the center of the sphere on which they are arranged is located outside the user's head.

First Embodiment
FIG. 4 is a functional block diagram of the noise suppression system according to the first embodiment, and FIG. 5 shows the processing flow thereof.

The noise suppression system includes a microphone array consisting of a reference microphone 91, a cancellation speaker 92, and L error microphones 93-i, a suppression signal generation unit 110, and a sound pressure estimation unit 120. Let i=1, 2, ..., L. The device consisting of the suppression signal generation unit 110 and the sound pressure estimation unit 120 is also called a suppression signal generation device.

The suppression signal generating device receives the pickup signal x(r) of the reference microphone 91 and the pickup signals x(e) of the L error microphones 93-i as input, generates a cancellation signal (hereinafter also referred to as the "suppression signal") y so that the point at which the amount of noise suppression is maximum is located closer to the user than the installation position of the error microphone 93-i, and outputs the generated signal to the cancellation speaker 92.

The suppression signal generating device is a special device configured by loading a special program into a publicly known or dedicated computer having, for example, a central processing unit (CPU), a main memory (RAM), etc. The suppression signal generating device executes each process under the control of the central processing unit, for example. Data input to the suppression signal generating device and data obtained by each process are stored in, for example, the main memory, and the data stored in the main memory is read out to the central processing unit as necessary and used for other processes. At least a part of each processing unit of the suppression signal generating device may be configured by hardware such as an integrated circuit. Each storage unit of the suppression signal generating device may 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 inside the suppression signal generating device, and may be configured by an auxiliary storage device configured by a semiconductor memory element such as a hard disk, optical disk, or flash memory, and may be configured to be provided outside the suppression signal generating device.

Each part is explained below.

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

<Cancellation Speaker 92>
The cancellation speaker 92 receives the cancellation signal y as input and reproduces the cancellation signal y (S92). When the playback sound reproduced from the cancellation speaker 92 and the noise to be suppressed are in completely opposite phase, the playback sound and the noise to be suppressed overlap, that is, the sound waves overlap, and the waves cancel each other out, thereby suppressing the noise.

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

<Sound Pressure Estimation Unit 120>
The sound pressure estimation unit 120 receives the output signals (sound pickup signals) x(e) of the L error microphones 93-i as input, calculates an estimated sound pickup signal x(v) which is a signal estimated to be picked up when the microphone 130 is installed at a position closer to the observation point than the L error microphones 93-i, and outputs the signal. That is, the sound pressure estimation unit 120 estimates a sound pickup signal obtained when a sound that is not suppressed by the sound reproduced from the cancellation speaker 92 is picked up at the installation position of the microphone 130 (S120), and outputs the estimated sound pickup signal as an estimated sound pickup signal x(v). Seven examples of estimation methods for the estimated sound pickup signal x(v) are given below. 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 virtual error microphone pickup signal is estimated from the pickup signals of multiple error microphones arranged on a spherical surface close to the user's head, using spherical harmonic function expansion coefficients. The conditions for estimation method 1 are described below.

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

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

(Condition 3)
The positions of the virtual microphones are the positions of the user's ears and are 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 extrapolation of the sound pressure.

Figures 6 and 7 are diagrams for explaining the positional relationship of multiple error microphones 93-i on the spherical surface of 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 ears, and multiple error microphones are placed at equal intervals (at the same radius) from the position of origin O'. That is, the error microphones are placed on the surface of virtual sphere B. Error microphones are placed on the spherical surface of virtual sphere B of radius a, and the sound pressure on the spherical surface of radius r is estimated. For example, the distance from the center (origin O') to the error microphone is a = 7.5 cm, and four error microphones are placed at the vertices of a triangular pyramid inscribed in the virtual sphere of radius a. In this case, the four error microphones form a tetrahedral 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 sphere from the observed sound pressure on a certain sphere.

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

Here, k is the wave number, β _n,m are the acoustic field coefficients specific to the noise field, j _n is the nth order spherical Bessel function, and Y ^m _n (·) is the spherical harmonic function.

The sound field coefficients β _n,m are calculated using spherical harmonic expansion.

It can be calculated as:

Therefore, the sound pressure p(r,Ω) at any angle Ω of radius r, excluding the noise source, is

It can be estimated by:

Therefore, the sound pressure estimation unit 120 uses the observed sound pressure values [p(a,Ω _{1), p(a,Ω 2} ), ..., p(a,Ω _L )] obtained from the output signals (picked up signals) x(e) of the _L error microphones 93-i to calculate the sound field coefficients β _n,m according to equation (2), and estimates the sound pressure p(r,Ω) at the virtual microphone according to equation (3) using the sound field coefficients β _n,m . In order to reduce the amount of calculations, sound field coefficients β _n,m corresponding to various observed sound pressure values may be obtained in advance, and combinations of the observed sound pressure values and the sound field coefficients β _{n,m may be stored in advance in a memory unit not shown.When estimating the sound pressure, the sound field coefficients β n,m} corresponding to the observed sound pressure values [p(a,Ω ₁ ), p(a,Ω ₂ ), ..., p(a,Ω _L )] obtained from the output signals (picked up sound signals) x(e) of the L error microphones 93 _-i may be retrieved from the memory unit, and the sound pressure at the virtual microphone may be estimated using equation (3). Furthermore, when 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 sound pressure values is obtained in advance, and combinations of the observed sound pressure values 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 are stored in advance in a memory unit not shown.When estimating the sound pressure, the sound pressure p(r, Ω) at the virtual microphone corresponding to the observed sound pressure values [p(a, Ω ₁ ), p(a, Ω ₂ ), ..., p(a, Ω _L )] obtained from the output signals (picked up signals) x(e) of the L error microphones 93-i is retrieved from the memory unit, and the sound pressure at the virtual microphone can be estimated without performing any calculation processing.

In addition, when 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 it is possible to express the frequency and radius in the range of kr<N with good accuracy. Here, r is the radius (distance to the virtual microphone), k=2πf/c (c is the speed of sound = 340m/s, f is frequency), so for example, when N=1, it is possible to express and predict frequencies up to 721Hz within a range of 7.5cm radius (distance to the virtual microphone). Since noise is concentrated in the low frequency range, it is quite useful if frequencies up to 721Hz can be predicted.

(Estimation method 2)
First, the conditions for estimation method 2 will be described.
The estimation method 2 also needs to satisfy the conditions 1 and 2 described in the estimation method 1. In the estimation method 2, the following condition 4 needs to be satisfied instead of the condition 3.

(Condition 4)
The virtual microphones are positioned at the positions of the user's ears and on the surface of the virtual sphere.

Therefore, in this estimation method, as in estimation method 1, the origin O' is placed at a position at a predetermined distance from the listener's ear position, and multiple error microphones are placed at equal intervals (same radius positions) from the origin position. That is, the error microphones are placed on the surface of a virtual sphere. Furthermore, multiple error microphones are placed 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). In this case, 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 can be obtained by the following equation.

The sound pressure estimation unit 120 obtains a sound field coefficient Pn,m according to equation (4) using the observed sound pressure values [p(a, _Ω1 ),p(a, _Ω2 ),...,p(a, _ΩL )] obtained from the output signals (collected sound signals) x(e) of the L error microphones 93 _-i , and estimates the sound pressure p(r,Ω) at the virtual microphone according to equation (5) using the sound field coefficient Pn _{, m} . Note that a configuration for reducing the amount of calculation described in estimation method 1 may be adopted by using the sound field coefficient Pn _{,m instead of the sound field coefficient βn,m} .

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

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

(Condition 6)
The position of the virtual microphone is set closer to the center of the virtual sphere than the position of the user's ear, and is set 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 (at the same radius). Furthermore, multiple error microphones are placed at equal intervals on the spherical surface of imaginary sphere B. For example, error microphones can be placed at equal intervals by placing them at the tangent points (each vertex of a regular polyhedron) between a regular polyhedron and its circumscribing sphere (outer radius a) or at the tangent points (center of each face of a regular polyhedron) between a regular polyhedron and its inscribing sphere (inscribed radius a). For example, error microphones are placed at each vertex of a regular tetrahedron inscribed in the imaginary sphere.

When 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 is (N+1) ² ≦L, and the sound field coefficient βn _,m can be calculated by the following formula, where the integral becomes a sum.

When predicting the sound pressure at the ear position using this sound field coefficient β _n,m , the distance r from the origin of the virtual sphere to the ear position is not used, but the sound pressure is estimated using the distance r _d <r inside the ear position.

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 coincide, but in this estimation method, the user's ear position and the virtual microphone position differ.

The sound pressure estimation unit 120 uses the observed sound pressure values [p(a,Ω ₁ ), p(a,Ω ₂ ), ..., p(a,Ω _L )] obtained from the output signals (sound pickup signals) x(e) of the L error microphones 93-i to calculate the sound field coefficient β _n,m according to equation (6), and estimates the sound pressure p(r d ,Ω) at the virtual microphone according to equation (7) using the sound field coefficient β _n,m . Note that the configuration for reducing the amount of calculation described in estimation method 1 may be adopted. However, sound pressure p(r _d ,Ω) is estimated instead of sound pressure p( _r ,Ω).

For example, this estimation method may be combined with estimation method 2 to change condition 6 to condition 6A as follows:

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

The virtual microphone is positioned inside the virtual sphere by (rr _d ) from the position of the user's ears. When the radius a of the virtual sphere is 7.5 cm, the prediction accuracy is equal to or higher than that of estimation method 1 by setting 0 < rr _d ≦ 6 (0 < rr _d ≦ 0.8a in terms of the ratio to the radius a of the virtual sphere), and the prediction accuracy is particularly high when rr _d = 3 (rr _d = 0.4a in terms of the ratio to the radius a of the virtual sphere).

(Estimation method 4)
First, the conditions for Estimation Method 4 will be described.
The estimation method 4 must also satisfy the conditions 1 and 3 described in the estimation method 1. The estimation method 4 does not need to satisfy the condition 2.

In this estimation method, only the three microphones on the surface of the sphere that are closest to the ears are used (see Figures 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, etc.

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

The sound pressure estimation unit 120 estimates spherical harmonic function expansion coefficients by the least squares method using the sound pressure observation values [p(a,Ω ₁ ),p(a,Ω ₂ ),...,p(a,Ω _L )] obtained from the output signals (sound pickup signals) x(e) of the L error microphones 93-i, and estimates the sound pressure p(r,Ω) at the virtual microphone using the spherical harmonic function expansion coefficients. Note that a configuration for reducing the amount of calculation described in estimation method 1 may be adopted by using spherical harmonic function expansion coefficients instead of the sound field coefficients β _n,m .

(Estimation method 5)
First, the conditions for Estimation Method 5 will be described.
The estimation method 5 also needs to satisfy the condition 2 described in the estimation method 1. The estimation method 5 also needs to satisfy the conditions 7 and 8 instead of the conditions 1 and 3.

(Condition 7)
The error microphones are arranged on the spherical surface of a hard sphere B' located close to the user's head. The hard sphere B' is a physical object that reflects sound (see FIG. 12).

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

In this estimation method, the sound pressure is estimated by assuming the sphere to be a hard sphere that reflects sound, rather than a virtual sphere.
By arranging multiple error microphones on the surface of the hard sphere B', the radius of the sphere can be made smaller. For example, in estimation method 1, if the radius a of the virtual sphere B is 7.5 cm, then by setting the radius a' of the hard sphere B' to 3 cm or more and less than 7.5 cm (in terms of the ratio to the radius a of the virtual sphere, 0.4a≦a'<1), it is possible to achieve prediction accuracy at least as good as that of estimation method 1. The prediction formula in this case is as follows:

Let a be the radius of the hard sphere, j _n ' be the derivative of j _n , h _n ⁽²⁾ be the second kind of spherical Hankel function, and h _n ' ⁽²⁾ be the derivative of h _n ⁽²⁾ .

The sound pressure estimation unit 120 uses the observed sound pressure values [p(a,Ω ₁ ), p(a,Ω ₂ ), ..., p(a,Ω _L )] obtained from the output signals (sound pickup signals) x(e) of the L error microphones 93-i to obtain the sound field coefficient B _n,m according to equation (9), and estimates the sound pressure p(r,Ω) at the virtual microphone according to equation (8) using the sound field coefficient B _{n, m} . Note that a 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} . This estimation method may also be combined with estimation method 3.

(Estimation method 6)
First, the conditions for Estimation Method 6 will be described.
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, estimation method 6 satisfies the following condition 9.

(Condition 9)
A directional microphone is used as the error microphone instead of an omnidirectional microphone.

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

The sound pressure estimation unit 120 uses the sound pressure observation values x[e]=[p(a,Ω ₁ ),p(a,Ω ₂ ),...,p(a,Ω _L )] from L error microphones (directional microphones) to obtain sound field coefficients β _n,m according to Ambisonics, and estimates the sound pressure p(r,Ω) at the virtual microphone using the sound field coefficients β _n,m . Note that the configurations described in estimation methods 1 to 5 for reducing the amount of calculation may be adopted.

(Estimation method 7)
First, the conditions for Estimation Method 7 will be described.
It is necessary to satisfy the conditions of Estimation Method 7 and Estimation Method 5. Furthermore, in Estimation Method 7, the following Condition 10 is satisfied.

(Condition 10)
At least one speaker is embedded in the rigid sphere B'.

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

By installing a speaker inside the hard sphere B', the requirements of Kirchhoff's integral equation (the sound source should not be located inside the radius to be extrapolated) can be satisfied when deriving the sound pressure prediction formula, and this is expected to have the effect of improving the extrapolation accuracy of the sound pressure.

The sound pressure estimation unit 120 uses the observed sound pressure values [p(a,Ω ₁ ),p(a,Ω ₂ ),...,p(a,Ω _L )] obtained from the output signals (sound pickup signals) x(e) of the L error microphones 93-i to calculate the sound field coefficient β _n,m according to equation (9), and estimates the sound pressure p(r,Ω) at the virtual microphone according to equation (8) using the sound field coefficient β _n,m . Note that the configuration for reducing the amount of calculation described in estimation method 1 may be adopted.

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

Conventional techniques can be used as a method for generating the cancellation signal. For example, the method described in Non-Patent Document 1 can be used. In this embodiment, a feedforward type ANC is realized by the collected sound signal x(r), the estimated collected sound signal x(v), and the cancellation signal y. The collected sound signal that would be obtained when the interference sound between the noise from the noise source and the reproduced sound of the cancellation signal y is detected by the virtual microphone 130 is estimated, and the noise from the noise source is detected by the reference microphone 91. The cancellation signal y is generated by inputting the noise from the noise source 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, which 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 by an adaptive algorithm so that the input to the virtual microphone 130 is minimized. Since a conventional update method can be used as a method for updating the coefficients of the noise control filter, a description will be omitted. In the feedforward type ANC, a secondary path model that estimates the secondary path is used to compensate for the influence of the secondary path in the adaptive algorithm.

＜Effects＞
With the above configuration, in cases where it is not possible to place an error microphone near the user's ear, high suppression performance can be achieved without placing multiple error microphones around the user's head so as to cover the head.

<Other Modifications>
The present invention is not limited to the above-mentioned embodiment and modified examples. For example, the above-mentioned various processes may be executed not only in chronological order as described, but also in parallel or individually depending on the processing capacity of the device executing the processes or as necessary. In addition, appropriate modifications are possible within the scope of the present invention.

<Program and recording medium>
The various processes described above can be implemented by loading a program that executes each step of the above method into the recording unit 2020 of the computer 2000 shown in FIG. 14, and operating the control unit 2010, input unit 2030, output unit 2040, display unit 2050, etc.

The program describing this processing can be recorded on a computer-readable recording medium. Examples of computer-readable recording media include magnetic recording devices, optical disks, magneto-optical recording media, and semiconductor memories.

The program may be distributed, for example, by selling, transferring, or lending portable recording media such as DVDs or CD-ROMs on which the program is recorded. Furthermore, the program may be distributed by storing the program in a storage device of a server computer and transferring the program from the server computer to other computers via a network.

A computer that executes such a program, for example, first stores in its own storage device the program recorded on a portable recording medium or the program transferred from a server computer. Then, when executing a process, the computer reads the program stored on its own recording medium and executes the process according to the read program. As another execution form of this program, the computer may read the program directly from the portable recording medium and execute the process according to the program, or may execute the process according to the received program each time a program is transferred from the server computer to this computer. In addition, the server computer may not transfer the program to this computer, but may execute the above-mentioned process by a so-called ASP (Application Service Provider) type service that realizes the processing function only by issuing an execution instruction and obtaining the results. Note that the program in this form includes information used for processing by an electronic computer that is equivalent to a program (such as data that is not a direct command to the computer but has properties that specify the processing of the computer).

In addition, in this embodiment, the device is configured by executing a specific program on a computer, but at least a portion of the processing may be realized by hardware.

Claims (8)

A generating device for generating a cancellation signal for use in active noise control, comprising:
generating a cancellation signal so that a 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 head of the user, and the center of the spherical surface is located outside the head of the user;
A sound pressure estimation unit that estimates a sound pickup signal that is picked up when a virtual microphone is installed at a position closer to the observation point than the plurality of error microphones by using a spherical harmonic function expansion coefficient, and obtains an estimated sound pickup signal x(v);
a suppression signal generation unit that generates a cancellation signal for suppressing noise at an installation position of the virtual microphone by using a pickup signal x(r) that picks up the noise to be suppressed and the estimated pickup signal x(v),
The virtual microphone is a microphone that is virtually installed without actually being installed.
generator. 2. The generating device of claim 1,
the plurality of error microphones are arranged on a spherical surface of a virtual sphere close to the head of the user ,
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 is on the surface or outside of the virtual sphere.
generator. 3. The generating device of claim 2,
The position of the virtual microphone is the position of the user's ear and is on the surface of the virtual sphere.
generator. 2. The generating device of claim 1,
the plurality of error microphones are disposed at equal intervals on a spherical surface of a virtual sphere close to the head of the user ,
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. 2. The generating device of claim 1,
The number of the error microphones is three;
The three error microphones are arranged on a spherical surface of a virtual sphere close to the user's head ,
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;
The sound pressure estimation unit estimates spherical harmonic function expansion coefficients by a least squares method, and obtains the estimated sound pickup signal x(v) using the estimated spherical harmonic function expansion coefficients.
generator. 2. The generating device of claim 1,
The plurality of error microphones are arranged on a 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 is outside the hard sphere.
generator. 1. A method for generating a cancellation signal for use in active noise control, comprising:
generating a cancellation signal so that a 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 head of the user, and the center of the spherical surface is located outside the head of the user;
a sound pressure estimation step of estimating a sound signal that would be picked up when a virtual microphone is placed at a position closer to the observation point than the plurality of error microphones, using spherical harmonic function expansion coefficients, to obtain an estimated sound pressure signal x(v);
a suppression signal generation step of generating a cancellation signal for suppressing noise at an installation position of the virtual microphone by using a pickup signal x(r) that picks up the noise to be suppressed and the estimated pickup signal x(v),
The virtual microphone is a microphone that is virtually installed without actually being installed.
Generation method. A program for causing a computer to function as a generating device according to any one of claims 1 to 6.