JP7675989B2 - GENERATION DEVICE, GENERATION METHOD, AND PROGRAM - Google Patents

Description

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.

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.

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 .

The present invention aims to provide a generation device, generation method, and program that estimates the sound signal that would be obtained if sound were picked up near the user's ear from the sound signal picked up at the actual error microphone placement position, and uses the estimated sound signal instead of the sound signal picked up at the actual error microphone placement position to actively control the cancellation signal, thereby achieving high suppression performance even near the user's ear away 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 to be used in active noise control. The generating device generates the cancellation signal so that the point at which the amount of noise suppression is maximized is located closer to the user than the installation position of the error microphone.

The present invention has the effect of achieving higher suppression performance than conventional techniques when an error microphone cannot be placed close to the user's ear.

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. 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 2. FIG. 13 is a diagram for explaining estimation method 3. FIG. 4 is a diagram showing a simulation result of the first embodiment. FIG. 13 is a diagram showing an example of the configuration of a computer to which the present technique is applied.

Hereinafter, an embodiment of the present invention will be described. 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 duplicated description will be omitted. In the following description, symbols such as "^" and " ^- " used in the text should be written directly above the character immediately following them, but due to limitations in text notation, they are written immediately before the character in question. In the formulas, these symbols are written in their original positions. Furthermore, unless otherwise specified, processing performed on each element of a vector or matrix is assumed to be applied to all elements of that vector or matrix.

<Key Points of the First Embodiment>
In this embodiment, the observed sound pressure at the ear is estimated from the pickup signal of the error microphone installed at a position away from the ear. For example, the pickup signal of a virtual error microphone placed at the ear is estimated from the pickup signal of the actual error microphone, and the pickup signal of the virtual error microphone is used as the pickup signal 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 a virtual error microphone. For example, the sound pressure is estimated by taking into account the distance attenuation and phase delay from the actual error microphone's installation position to the ear. In addition, for example, the sound pressure at the ear is estimated using spherical harmonics from an actual error microphone placed on a sphere.

First Embodiment
FIG. 3 is a functional block diagram of the noise suppression system according to the first embodiment, and FIG. 4 shows the processing flow thereof.

The noise suppression system includes a reference microphone 91, a cancellation speaker 92, an error microphone 93, a suppression signal generation unit 110, and a sound pressure estimation unit 120. 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 signal x(e) of the error microphone 93 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, and outputs the cancellation 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>
The error microphone 93 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 is placed at a position closer to the noise source than the observation point (e.g., the user's ear). For example, the error microphone 93 is placed at a position 0.05 m closer to the noise source than the user's ear, as shown in FIG. 5.

<Sound Pressure Estimation Unit 120>
The sound pressure estimation unit 120 receives the output signal (sound pickup signal) x(e) of the error microphone 93 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 error microphone 93, 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). Three 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 pickup signal x(v) of the virtual microphone 130 is estimated from an actual pickup signal x(e) of the error microphone 93 based on distance attenuation and phase delay from the error microphone 93 and the virtual microphone 130. Fig. 5 is a diagram for explaining the positional relationship between the noise source, the error microphone 93, and the virtual microphone 130.

In this estimation method, the position of a noise source is assumed, and it is assumed that noise propagates as a plane wave from the noise source to the error microphone 93 and the virtual microphone 130. The sound signal picked up by the virtual microphone 130 is estimated by estimating the distance attenuation and phase shift from the error microphone 93 to the observation point from the transfer function from the noise source to the error microphone 93 and the transfer function from the noise source to the observation point (position of the virtual microphone 130). The sound pressure estimation unit 120 estimates the sound signal picked up by the virtual microphone 130 from the output signal (sound pickup signal) x(e) of the error microphone 93 by the following equation, and outputs the estimated sound pickup signal x(v)=[^G _p1 ^G _p2 ].

^G _pn =w _n x(e) (n=1,2) (1)
Here, if we only consider the gain attenuation, _wn is
_wn = | _Gpn |/| _Ge | (2)
If we only consider the phase shift, _wn is
w _n =exp((arg G _pn -arg G _e )j) (3)
G _e and G _pn in formulas (2) and (3) are calculated in advance from the assumed noise source position and observation point prior to estimation processing. For example, a loudspeaker for the noise source is placed at the assumed noise source position, a predetermined signal is played back by the loudspeaker for the noise source, G _e is obtained from a signal picked up by a microphone placed at the error microphone position, and G _pn is obtained from a signal picked up by a microphone placed at the observation point position.
(Estimation method 2)
In this estimation method, a signal picked up by a virtual error microphone is estimated from signals picked up by multiple error microphones arranged at equal intervals near the head, using spherical harmonic expansion coefficients. Fig. 6 is a diagram for explaining the positional relationship of the actual error microphones.

In this estimation method, error microphones are arranged at equal intervals on a sphere of radius r _e , and the sound pressure on the sphere of radius r is estimated. For example, the distance from the center to the error microphones is set to r _e =0.15 m, and the error microphones can be arranged at equal intervals by (i) arranging six error microphones at the center of each face of a regular hexahedron (see (i) of FIG. 6 ), or (ii) arranging twelve error microphones at the center of each face of a regular dodecahedron (see (ii) of FIG. 6 ). For example, the distance from the center to the observation point (the position of the virtual microphone 130) is estimated as r=0.08 m.

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.

Observation values of sound pressure p( _θ1 , _φ1 ),p( _θ2 , _φ2 ),...,p( _θL , _φL ) are obtained from L error microphones on a radius r _e . For example, the sound pickup signals x(e) of the L error microphones 93 are set to [p( _θ1 , _φ1 ),p( _θ2 , _φ2 ),...,p( _θL , _φL )].

音圧推定部１２０は、求めた音場係数P_nm(r_e)を用いて、次式により、半径r上の音場係数P_nm(r)を求める。

ただし、kは波数であり、aは音を反射する剛球の半径とし、j_nはn次の球面ベッセル関数であり、j_n'はj_nの微分であり、h_n ⁽²⁾は第二種球ハンケル関数であり、h_n'⁽²⁾はh_n ⁽²⁾の微分である。
音圧推定部１２０は、再合成により、観測点(r,θ,φ)における音圧の推定値^p(r,θ,φ)を得る。

なお、推定収音信号x(v)=^p(r,θ,φ)とする。 The sound pressure estimation unit 120 obtains a sound field coefficient P _nm (r _e ) on a radius r _e for the spherical harmonic function Y ^m _n (·) by the following equation.

The sound pressure estimation unit 120 uses the obtained sound field coefficient P _nm (r _e ) to obtain a sound field coefficient P _nm (r) on a radius r according to the following equation.

where k is the wave number, a is the radius of the hard sphere that reflects sound, j _n is the nth order spherical Bessel function, j _n ' is the derivative of j _n , h _n ⁽²⁾ is the second kind of spherical Hankel function, and h _n ' ⁽²⁾ is the derivative of h _n ⁽²⁾ .
The sound pressure estimation unit 120 obtains an estimate value ^p(r, θ, φ) of the sound pressure at the observation point (r, θ, φ) by resynthesis.

The estimated picked-up signal is x(v)=^p(r,θ,φ).

The derivation of equation (5) is explained below.

である。なお、B_nmは騒音源の座標と信号で定まる係数である。球面調和関数展開は、

であり、音場係数P_nm(r)、音場係数P_nm(r_e)は、次式で表される。

式(10)をB_nm=…の形に変形して(9)に代入すると、

となる。

なお、球面調和関数展開における最大次数Nは以下の制約を受ける。 When the noise source is a point source and reflection from a hard sphere of radius a is taken into consideration, the sound pressure at point (r, θ, φ) is

Note that B _nm is a coefficient determined by the coordinates of the noise source and the signal. The spherical harmonic expansion is

and the sound field coefficients P _nm (r) and P _nm (r _e ) are expressed by the following equations.

Transforming equation (10) into the form B _nm = ... and substituting it into (9), we get

It becomes.

The maximum order N in the spherical harmonic expansion is subject to the following constraints:

(N+1) ² <L
Here, the number of speakers required corresponds to each mode of the spherical harmonic function Y ^m _n (·). If L=6, then N=1, and if L=12, then N=2.

In addition, to prevent spatial aliasing, N is subject to the following constraints:

kr<N
The distance between the head and the virtual error microphone is limited. For example, when the frequency is 300 Hz and N=1, the estimation area is limited to within the distance r=0.18 m from the head.

(Estimation method 3)
In this estimation method, the signal picked up by a virtual error microphone is estimated from the signal picked up by a plurality of error microphones arranged at non-uniform intervals near the head, using spherical harmonic expansion coefficients estimated by the least squares method. FIG. 7 is a diagram for explaining the positional relationship of the error microphones. For example, error microphones are arranged at four points behind the head (four azimuth angles (0°, 30°, 150°, 180°) × elevation angle 0°) or at 12 points behind the head (four azimuth angles (0°, 30°, 150°, 180°) × three elevation angles (-30°, 0°, 30°)). The installation radius of the error microphones may be determined according to the environment in which they are installed and the type of sound to be suppressed. For example, when used in a train where running noise is to be suppressed, the installation radius of the error microphones is set to 0.13 m, taking into account the seat size.

In this estimation method, error microphones are placed at non-uniform intervals on a sphere of radius r _e , 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.

なお、ω_i＝(θ_i,φ_i)とし、i=1,2,…,Lとする。 Observation values of sound pressure p(r _e ,θ ₁ ,φ ₁ ),p(r _e ,θ ₂ ,φ ₂ ),...,p(r _e ,θ _L ,φ _L ) are obtained from _L error microphones on a radius of r e . The sound pickup signal of the error microphone 93 is defined as x(e)= ^- p=[p(r _e ,θ ₁ ,φ ₁ ),p(r _e ,θ ₂ ,φ ₂ ),...,p(r _e ,θ _L ,φ _L )] ^{T. -p} is expressed as follows.

Here, ω _{i =} (θ _i , φ _i ), where i = 1, 2, ..., L.

推定したい点が半径r上にあるとしたとき、展開係数P_nm(r)および推定値の計算は推定方法２と同様である。つまり、
音圧推定部１２０は、次式により、球面調和関数Y^m _n(・)に対する半径r_e上の音場係数P_nm(r_e)を求める。

音圧推定部１２０は、求めた音場係数P_nm(r_e)を用いて、次式により、半径r上の音場係数P_nm(r)を求める。

音圧推定部１２０は、再合成により、観測点(r,θ,φ)における音圧の推定値^p(r,θ,φ)を得る。

なお、推定収音信号x(v)=^p(r,θ,φ)とする。 The sound pressure estimation unit 120 obtains, as a solution, ^−P (r _e ) that minimizes the absolute square error.

When the point to be estimated is on the radius r, the calculation of the expansion coefficients P _nm (r) and the estimated value is the same as in Estimation Method 2. That is,
The sound pressure estimation unit 120 obtains a sound field coefficient P _nm (r _e ) on a radius r _e for the spherical harmonic function Y ^m _n (·) by the following equation.

The sound pressure estimation unit 120 uses the obtained sound field coefficient P _nm (r _e ) to obtain a sound field coefficient P _nm (r) on a radius r according to the following equation.

The sound pressure estimation unit 120 obtains an estimate ^p(r, θ, φ) of the sound pressure at the observation point (r, θ, φ) by resynthesis.

The estimated picked-up signal is x(v)=^p(r,θ,φ).

<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, it is possible to achieve higher suppression performance than in the past when it is not possible to place an error microphone close to the user's ear. Figure 8 shows the simulation results of the first embodiment. In (A), the noise is a 300 Hz plane wave, and in (B), the noise is a 100 Hz plane wave.

<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 memory unit 2020 of the computer shown in Figure 9, and operating the control unit 2010, input unit 2030, output unit 2040, 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 (7)

A generating device for generating a cancellation signal for use in 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 position of the error microphone;
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 error microphone, 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,
The sound pressure estimation unit obtains the estimated sound pickup signal x(v) from the sound pickup signals x(e) of the plurality of error microphones arranged at equal intervals near the user's head by using spherical harmonic function expansion coefficients.
generator. A generating device for generating a cancellation signal for use in 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 position of the error microphone;
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 error microphone, 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,
The sound pressure estimation unit estimates a sound pressure from the sound signal x(e) of the error microphone based on distance attenuation and phase delay from the positional relationship between the error microphone and the virtual microphone.
x(v)=[^G _p1 ^G _p2 ]
^G _pn =w _n x(e) (n=1,2)
The estimated sound pickup signal x(v) is obtained by the above. The transfer characteristic from the assumed noise source position to the error microphone position is Ge _, and the transfer characteristic from the assumed noise source position to the observation point position is Gpn _. When only the gain attenuation is taken into consideration, _wn is
w _n =|G _pn |/|G _e |
If we only consider the phase shift, _wn is
w _n =exp((arg G _pn -arg G _e )j)
That is,
generator. A generating device for generating a cancellation signal for use in 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 position of the error microphone;
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 error microphone, 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,
The sound pressure estimation unit obtains the estimated sound pickup signal x(v) from the sound pickup signals x(e) of the plurality of error microphones arranged at non-equidistant intervals near the user's head by using spherical harmonic function expansion coefficients estimated by a least squares method.
generator. 1. A method for generating a cancellation signal for use in 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 position of the error microphone;
a sound pressure estimation step of estimating a sound pickup signal that would be picked up if a virtual microphone were installed at a position closer to the observation point than the error microphone, and obtaining an estimated sound pickup 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,
The sound pressure estimation step obtains the estimated sound pickup signal x(v) from the sound pickup signals x(e) of the plurality of error microphones arranged at equal intervals near the user's head by using spherical harmonic function expansion coefficients.
Generation method. 1. A method for generating a cancellation signal for use in 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 position of the error microphone;
a sound pressure estimation step of estimating a sound pickup signal that would be picked up if a virtual microphone were installed at a position closer to the observation point than the error microphone, and obtaining an estimated sound pickup 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,
The sound pressure estimation step is to estimate a sound pressure from the sound signal x(e) of the error microphone based on distance attenuation and phase delay from the positional relationship between the error microphone and the virtual microphone.
x(v)=[^G _p1 ^G _p2 ]
^G _pn =w _n x(e) (n=1,2)
The estimated sound pickup signal x(v) is obtained by the above. The transfer characteristic from the assumed noise source position to the error microphone position is Ge _, and the transfer characteristic from the assumed noise source position to the observation point position is Gpn _. When only the gain attenuation is taken into consideration, _wn is
w _n =|G _pn |/|G _e |
If we only consider the phase shift, _wn is
w _n =exp((arg G _pn -arg G _e )j)
That is,
Generation method. 1. A method for generating a cancellation signal for use in 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 position of the error microphone;
a sound pressure estimation step of estimating a sound pickup signal that would be picked up if a virtual microphone were installed at a position closer to the observation point than the error microphone, and obtaining an estimated sound pickup 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,
The sound pressure estimation step obtains the estimated sound pickup signal x(v) from the sound pickup signals x(e) of the plurality of error microphones arranged at non-equidistant intervals near the user's head by using spherical harmonic function expansion coefficients estimated by a least squares method.
Generation method. A program for causing a computer to function as the generating device according to any one of claims 1 to 3 .

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