Generating device, generating method, and program

By estimating sound pressure using spherical harmonic functions to simulate a virtual microphone, the system enhances noise cancellation near the user's ear, overcoming the limitations of conventional systems with error microphones.

JP7709142B2Active Publication Date: 2025-07-16NIPPON TELEGRAPH & TELEPHONE CORP +1
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Patent Information

Application Number
JP2022129835
Authority / Receiving Office
JP · JP
Patent Type
Patents
Current Assignee / Owner
Filing Date
2022-08-17
Publication Date
2025-07-16
Estimated Expiration
2042-08-17

AI Technical Summary

Technical Problem

Existing active noise control systems face reduced suppression performance when error microphones cannot be installed near the user's ear, leading to ineffective noise cancellation.

Method used

A generation device estimates a sound collection signal using spherical harmonic function expansion coefficients to simulate a virtual microphone positioned closer to the user's ear, generating a cancellation signal to maximize noise suppression at this point.

Benefits of technology

Achieves high suppression performance without the need for multiple error microphones around the head, effectively canceling noise near the user's ear.

✦ Generated by Eureka AI based on patent content.

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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
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Description

Technical Field

[0001] The present invention relates to an active noise control (ANC) technology for suppressing external noise at a specific position.

Background Art

[0002] Non-Patent Document 1 is known as a conventional active noise control technology. In active noise control, a reference microphone, an error microphone, and a cancellation speaker are generally used. FIG. 1 shows a configuration example of a conventional noise suppression device. The reference microphone 91 picks up the noise emitted from the noise source. The cancellation speaker 92 plays back 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 and feeds back the remaining noise. The suppression signal generation device 90 actively controls and generates a cancellation signal using the sound pickup signal of the reference microphone 91 and the sound pickup signal of the error microphone 93 so that the remaining noise is reduced. At the installation position of the error microphone 93, the cancellation speaker 92 emits a cancellation sound so that the remaining noise is reduced. Therefore, the cancellation sound suppresses the noise most efficiently at the installation position of the error microphone 93. Therefore, the error microphone 93 is installed near the user's ear.

[0003] However, in actual use, there may be cases where the error microphone 93 cannot be installed near the user's ear. When the distance between the installation position of the error microphone 93 and the user's ear increases, as described above, noise is most efficiently suppressed at the installation position of the error microphone 93, and the remaining noise at the user's ear becomes large, resulting in a decrease in the suppression performance, and the user may not be able to fully enjoy the benefits of noise suppression. For example, when the distance from the noise source to the ear is 100 mm, the suppression performance when the error microphone 93 is installed at the user's ear (0 mm) is -∞ dB, and the suppression performance when the error microphone 93 is installed at the midpoint between the noise source and the ear is -7.38 dB, which was confirmed by simulation. FIG. 2 is a diagram for explaining the difference between the suppressible region (sweet spot) S1 of the prior art and the desired sweet spot S2.

[0004] 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 ear sound pressure is predicted by estimating the internal sound pressure from the signals obtained by the error microphones 93.

Prior Art Documents

Non-Patent Documents

[0005]

Non-Patent Document 1

Non-Patent Document 2

Summary of the Invention

Problems to be Solved by the Invention

[0006] However, considering the use in seats of public institutions, etc., there is a problem that the installation conditions of a plurality of error microphones are not realistic. The present invention aims to provide a generation device, a generation method, and a program thereof, which estimate a sound collection signal obtained when collecting sound near the user's ear from a sound collection signal collected by an error microphone installed based on installation conditions suitable for use in seats of public institutions, etc., and actively control a cancellation signal. By using the estimated sound collection signal instead of the sound collection signal collected at the actual installation position of the error microphone, high suppression performance can be achieved even near the user's ear away from the actual error microphone.

Means for Solving the Problems

[0007] 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 a point where the noise suppression amount is maximized is located closer to the user than the installation positions of a plurality of error microphones. The plurality of error microphones are arranged on a spherical surface close to the user's head, and using spherical harmonic function expansion coefficients, a sound pressure estimation unit estimates a sound collection signal that would be collected when a virtual microphone is installed at a position closer to the observation point than the plurality of error microphones, and obtains an estimated sound collection signal x(v). The generation device includes a suppression signal generation unit that generates a cancellation signal for suppressing noise at the installation position of the virtual microphone using the sound collection signal x(r) that collects the noise to be suppressed and the estimated sound collection signal x(v). The virtual microphone is a microphone that is virtually installed without being actually installed.

Effects of the Invention

[0008] According to the present invention, when it is not possible to arrange an error microphone near the user's ear, it is possible to achieve high suppression performance without installing a plurality of error microphones so as to cover the head around the user's head.

Brief Description of the Drawings

[0009]

Figure 1

Figure 2

Figure 3

Figure 4

Figure 5

Figure 6

Figure 7

Figure 8

Figure 9

Figure 10

Figure 11

Figure 12

Figure 13

Figure 14

Embodiments for Carrying Out the Invention

[0010] Hereinafter, embodiments of the present invention will be described. In the drawings used in the following description, components having the same function and steps performing the same process are denoted by the same reference numerals, and redundant explanations are omitted. In the following description, the processing performed on each element unit of a vector or matrix is applied to all elements of that vector or matrix unless otherwise specified.

[0011] <Highlights of the First Embodiment> In this embodiment, the observed sound pressure near the user's ear is estimated from the sound collection signals of a plurality of error microphones installed on the spherical surface close to the user's head. For example, from the sound collection signals of the actual plurality of error microphones, the sound collection signal of a virtual error microphone arranged near the ear is estimated, and in ANC, the sound collection signal of the virtual error microphone is used as the sound collection signal of the conventional error microphone. By adopting such a 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 for canceling the remaining sound near the ear can be emitted.

[0012] As a method for estimating the sound collection signal of the virtual error microphone, various methods can be considered. For example, the sound pressure near the ear is estimated using spherical harmonic functions from the actual plurality of error microphones arranged on the spherical surface close to the user's head. Note that the plurality of error microphones are arranged such that the center of the spherical surface on which the plurality of error microphones are arranged is located outside the user's head.

[0013] <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.

[0014] The noise suppression system includes a reference microphone 91, a cancellation speaker 92, a microphone array composed of 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 composed of the suppression signal generation unit 110 and the sound pressure estimation unit 120 is also referred to as a suppression signal generation device.

[0015] The suppression signal generation device takes the sound collection signal x(r) of the reference microphone 91 and the sound collection signals x(e) of the L error microphones 93-i as inputs, and generates a cancellation signal (hereinafter also referred to as a "suppression signal") y such that the point where the noise suppression amount is maximized is located closer to the user side than the installation position of the error microphone 93-i, and outputs it to the cancellation speaker 92.

[0016] The suppression signal generation device is a special device configured by loading a special program into a known or dedicated computer having, for example, a central processing unit (CPU) and a main memory device (RAM: Random Access Memory). The suppression signal generation device executes each process under the control of, for example, the central processing unit. The data input to the suppression signal generation device and the data obtained by each process are stored, for example, in the main memory device, and the data stored in the main memory device is read out to the central processing unit as needed and used for other processes. At least a part of each processing unit of the suppression signal generation device may be configured 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 memory 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 generation device, and may be configured by an auxiliary storage device configured by a semiconductor memory element such as a hard disk, an optical disk, or a flash memory (Flash Memory), and may be provided outside the suppression signal generation device.

[0017] Hereinafter, each part will be described.

[0018] <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 picked up by the reference microphone 91 is hereinafter referred to as "noise".

[0019] <Cancellation speaker 92> The cancellation speaker 92 takes the cancellation signal y as an input and reproduces the cancellation signal y (S92). When the reproduced sound from the cancellation speaker 92 and the noise to be suppressed are in a completely opposite phase, the reproduced sound and the noise to be suppressed overlap, that is, the sound waves overlap, and the waves cancel each other out, so the noise is suppressed.

[0020] <Error microphone 93-i> The error microphone 93-i picks up the sound that was not suppressed by the playback sound reproduced from the cancellation speaker 92, including the remaining noise (S93), and outputs a pickup signal x(e). The error microphone 93-i is arranged at a position closer to the noise source than the observation point (for example, near the user's ear). Since the positions of the L error microphones 93-i are related to the sound pressure estimation method, they will be described together with the sound pressure estimation unit 120 described later.

[0021] <Sound pressure estimation unit 120> The sound pressure estimation unit 120 takes as input the output signals (pickup signals) x(e) of the L error microphones 93-i, and calculates and outputs an estimated pickup signal x(v), which is a signal estimated to be picked up when a microphone 130 is installed at a position closer to the observation point than the L error microphones 93-i. That is, the sound pressure estimation unit 120 estimates the pickup signal obtained when the sound not suppressed by the playback sound reproduced from the cancellation speaker 92 is picked up at the installation position of the microphone 130 (S120), and outputs the estimated pickup signal as the estimated pickup signal x(v). Hereinafter, seven examples of the estimation method of the estimated pickup signal x(v) will be illustrated. Here, the microphone 130 is virtually installed without being actually installed, and will be hereinafter referred to as the virtual microphone 130.

[0022] (Estimation method 1) In this estimation method, the pickup signal of a virtual error microphone is estimated from the pickup signals of a plurality of error microphones arranged on the spherical surface close to the user's head by using the spherical harmonic expansion coefficients. Hereinafter, the conditions of Estimation method 1 will be described.

[0023] (Condition 1) The plurality of error microphones are arranged on the spherical surface of a virtual sphere (hereinafter, also referred to as the virtual sphere B) close to the user's head. The virtual sphere is not a physically existing sphere. Further, the plurality of error microphones are arranged such that the center of the virtual sphere B is located outside the user's head.

[0024] (Condition 2) In order to use the spherical harmonic expansion coefficients, the number L of error microphones and the expansion order N of the spherical harmonic functions are (N + 1)2 satisfies ≤ L.

[0025] (Condition 3) The position of the virtual microphone is set as 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 extrapolation of sound pressure.

[0026] FIG. 6 and FIG. 7 are diagrams for explaining the positional relationship of a plurality of error microphones 93-i on the spherical surface of the virtual sphere B close to the user's head.

[0027] In this estimation method, an origin O' is placed at a position at a predetermined distance from the position of the listener's ear, and a plurality of error microphones are arranged at equally spaced positions (positions with the same radius) from the position of the origin O'. That is, error microphones are arranged on the surface of the virtual sphere B. Error microphones are arranged on the spherical surface of the virtual sphere B with a radius a, and the sound pressure on the spherical surface with a 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 arranged at the vertices of a triangular pyramid inscribed in the virtual sphere with a radius a. At this time, the four error microphones form a tetrahedral microphone array. For example, the distance from the center (origin O') to the observation point (the position of the virtual microphone 130) is set to r = 8 cm for estimation.

[0028] By using spherical harmonic function expansion, it is possible to estimate the observed sound pressure on an arbitrary spherical surface from the observed sound pressure on a certain spherical surface.

[0029] A certain angle Ω on the surface of a sphere with a radius a o =(θ o , φ o ) at which the observed sound pressure p(a, Ω o ) is, when noise arrives only from the outside of the sphere,

Equation

[0030] Sound field coefficient β n,m is calculated using spherical harmonic expansion

Number

[0031] Therefore, the sound pressure p(r,Ω) at an arbitrary angle Ω with radius r without including ambient noise sources is

Number

[0032] Accordingly, the sound pressure estimation unit 120 uses the observed values of the sound pressure [p(a,Ω1), p(a,Ω2),…, p(a,Ω L )] obtained from the output signals (acoustic signals) x(e) of the L error microphones 93-i, and obtains the sound field coefficient β n,m by Equation (2), and estimates the sound pressure p(r,Ω) at the virtual microphone by Equation (3) using the sound field coefficient β n,m . In order to reduce the computational amount, the sound field coefficient β n,m corresponding to various observed values of the sound pressure is obtained in advance, and the combination of the observed value of the sound pressure and the sound field coefficient β n,m is stored in advance in a storage unit (not shown). When estimating the sound pressure, the observed value of the sound pressure [p(a,Ω1), p(a,Ω2),…, p(a,Ω L )] obtained from the output signals (acoustic signals) x(e) of the L error microphones 93-i is retrieved from the storage unit, and the sound pressure at the virtual microphone is estimated by Equation (3) using the corresponding sound field coefficient β n,m . 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 values of the sound pressure is obtained in advance, and the observed value of the sound pressure x[e]=[p(a,Ω1), p(a,Ω2),…, p(a,Ω L)]The combination with the sound pressure p(r, Ω) in the virtual microphone is pre-stored in a storage unit (not shown), and when estimating the sound pressure, the observed values of the sound pressure [p(a, Ω1), p(a, Ω2), …, p(a, Ω L )]corresponding to ]] are retrieved for the sound pressure p(r, Ω) in the virtual microphone corresponding thereto, and the sound pressure in the virtual microphone may be estimated without performing arithmetic processing.

[0033] Note that when L = 4, (N + 1) 2 N is linearly restricted in order to satisfy ≦ L. On the other hand, when expressing the sound field using spherical harmonic expansion, it is known that the frequency and radius in the range up to kr < N can be accurately expressed. 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, when N = 1, if it is within the range of a radius (distance to the virtual microphone) of 7.5 cm, it can be seen that frequencies up to 721 Hz can be expressed and predicted. Since noise is concentrated in the low-frequency range, if frequencies up to 721 Hz can be predicted, it has sufficient utility value.

[0034] (Estimation method 2) First, the conditions of estimation method 2 will be described. 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 the following condition 4.

[0035] (Condition 4) The position of the virtual microphone is the position of the user's ear and is the surface of the virtual sphere.

[0036] Therefore, in this estimation method, similar to Estimation Method 1, the origin O’ is placed at a position at a predetermined distance from the listener's ear position, and a plurality of error microphones are arranged at equally spaced positions (positions with the same radius) from the origin position. That is, the error microphones are arranged on the surface of the virtual sphere. Further, a plurality of error microphones are arranged such 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), since r and a are the same, the division of the spherical Bessel function j n (·) disappears, and the estimated sound pressure can be obtained by the following equation.

[0037] [Number] [Number]

[0038] The sound pressure estimation unit 120 obtains the observed values of the sound pressure [p(a,Ω1), p(a,Ω2), …, p(a,Ω L )] obtained from the output signals (acoustic signals) x(e) of the L error microphones 93-i, and obtains the sound field coefficient P n,m using Equation (4), and estimates the sound pressure p(r,Ω) at the virtual microphone using Equation (5) with the sound field coefficient P n,m . Note that, instead of the sound field coefficient β n,m , a configuration for reducing the calculation amount described in Estimation Method 1 may be adopted using the sound field coefficient P n,m .

[0039] (Estimation Method 3) First, the conditions of Estimation Method 3 will be described. Estimation Method 3 also needs to satisfy Condition 2 described in Estimation Method 1. In Estimation Method 3, instead of Conditions 1 and 3, the following Conditions 5 and 6 need to be satisfied.

[0040] (Condition 5) A plurality of error microphones are arranged at equal intervals on the spherical surface of the virtual sphere B close to the user's head.

[0041] (Condition 6) The position of the virtual microphone is set to be 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 interpolation of sound pressure.

[0042] Therefore, in this estimation method, similar to Estimation Method 1, an origin is set at a position at a predetermined distance from the ear position of the listener, and a plurality of error microphones are arranged at equally spaced positions (positions with the same radius) from the origin position. Further, the plurality of error microphones are arranged at equal intervals on the spherical surface of the virtual sphere B. For example, error microphones can be arranged at the contact points (each vertex of the regular polyhedron) between a regular polyhedron and its circumscribed spherical surface (outer radius a), or at the contact points (the center of each face of the regular polyhedron) between a regular polyhedron and its inscribed spherical surface (inner radius a), thereby arranging the error microphones at equal intervals. For example, error microphones are arranged at each vertex of a regular tetrahedron inscribed in the virtual sphere.

[0043] When the number of error microphones is L and the L error microphones are arranged at equal intervals on the virtual spherical surface, the maximum order N of n is (N + 1) 2 ≦L, and the sound field coefficient β n,m is obtained by the following equation where the integration becomes a sum.

[0044] [Equation]

[0045] 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, the sound pressure is estimated using a distance r d < r that is closer to the inside. [Equation] FIG. 9 shows the relationship between the ear position of the user and the positions of the virtual sphere, error microphones, and virtual microphone. In Estimation Methods 1 and 2, the ear position of the user coincides with the position of the virtual microphone, but in this estimation method, the ear position of the user and the position of the virtual microphone are different.

[0046] The sound pressure estimation unit 120 uses the observed values of sound pressure [p(a,Ω1), p(a,Ω2),…, p(a,Ω L )] obtained from the output signals (acoustic signals) x(e) of the L error microphones 93-i, and obtains the sound field coefficient β according to Equation (6). n,m Using the sound field coefficient β n,m estimates the sound pressure p(r d ,Ω) at the virtual microphone according to Equation (7). Note that a configuration for reducing the computational complexity described in Estimation Method 1 may be adopted. However, instead of the sound pressure p(r,Ω), the sound pressure p(r d ,Ω) is estimated.

[0047] For example, by combining this estimation method and Estimation Method 2, Condition 6 may be changed to Condition 6A as follows.

[0048] (Condition 6A) Assume the position of the user's ear is on the surface of the virtual sphere, 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.

[0049] From the position of the user's ear, place the position of the virtual microphone inside the virtual sphere by (r - r d ). When the radius a of the virtual sphere is 7.5 cm, set 0 < r - r d ≤ 6 (in terms of the ratio to the radius a of the virtual sphere, 0 < r - r d ≤ 0.8a). In this way, the prediction accuracy is equal to or higher than that of Estimation Method 1. In particular, when r - r d = 3 (in terms of the ratio to the radius a of the virtual sphere, r - r d = 0.4a), the prediction accuracy is the highest.

[0050] (Estimation Method 4) First, the conditions of Estimation Method 4 will be described. Estimation Method 4 also needs to satisfy Conditions 1 and 3 described in Estimation Method 1. In Estimation Method 4, it is not necessary to satisfy Condition 2.

[0051] In this estimation method, only three microphones close to the ear among the microphones on the spherical surface are used (see FIGS. 10 and 11). In this case, since the condition 2 is not satisfied and the sound field coefficient cannot be obtained analytically, it is estimated using the least squares method or the like.

[0052] In this estimation method, error microphones are arranged at non-uniform intervals on the spherical surface with radius a, and the sound pressure on the spherical surface with radius r is estimated. In this estimation method, since the spherical harmonic function expansion cannot be directly used, the spherical harmonic function expansion coefficients are estimated by the least squares method to obtain the sound pressure on the spherical surface with radius r.

[0053] The sound pressure estimation unit 120 estimates the spherical harmonic function expansion coefficients by the least squares method using the observed values of the sound pressure [p(a,Ω1), p(a,Ω2),…, p(a,Ω L )] obtained from the output signals (acoustic 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, instead of the sound field coefficient β n,m , a configuration for reducing the computational amount described in Estimation Method 1 may be adopted using the spherical harmonic function expansion coefficients.

[0054] (Estimation Method 5) First, the conditions of Estimation Method 5 will be described. Estimation Method 5 also needs to satisfy Condition 2 described in Estimation Method 1. In Estimation Method 5, Conditions 7 and 8 are satisfied instead of Conditions 1 and 3.

[0055] (Condition 7) The plurality of error microphones are arranged on the spherical surface of the rigid sphere B' close to the user's head. The rigid sphere B' physically exists and is an object that reflects sound (see FIG. 12).

[0056] (Condition 8) The position of the virtual microphone is set as the position of the user's ear and is outside the virtual sphere.

[0057] In this estimation method, the sound pressure is estimated by using a rigid sphere that reflects sound instead of a virtual sphere. By arranging a plurality of 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, when the radius a of the virtual sphere B is 7.5 cm, by setting the radius a' of the rigid sphere B' to be 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 equal to or higher than that of Estimation Method 1. The prediction formula in this case is as follows.

[0058]

Number

Number

[0059] The sound pressure estimation unit 120 uses the observed values of the sound pressure [p(a, Ω1), p(a, Ω2), …, p(a, Ω L )] obtained from the output signals (acoustic signals) x(e) of the L error microphones 93-i to obtain the sound field coefficient B n,m by Equation (9), and estimates the sound pressure p(r, Ω) at the virtual microphone by Equation (8) using the sound field coefficient B n,m . Note that instead of the sound field coefficient β n,m , the sound field coefficient B n,m may be used, and a configuration for reducing the computational amount described in Estimation Method 1 may be adopted. Also, this estimation method and Estimation Method 3 may be combined.

[0060] (Estimation Method 6) First, the conditions of Estimation Method 6 will be described. Estimation Method 6 can be combined with any one of Estimation Methods 1 to 5, and Estimation Method 6 needs to satisfy the conditions of any one of Estimation Methods 1 to 5. Furthermore, Estimation Method 6 satisfies the following Condition 9.

[0061] (Condition 9) Use a directional microphone instead of an omnidirectional microphone as the error microphone.

[0062] By using a directional microphone, the radius of the virtual sphere B can be minimized as much as possible, and the direct sound field coefficient can be obtained and extrapolated according to ambisonics.

[0063] The sound pressure estimation unit 120 uses the observed values x[e] = [p(a,Ω1), p(a,Ω2), …, p(a,Ω L )] of the sound pressure from L error microphones (however, directional microphones) to obtain the sound field coefficient β n,m according to ambisonics, and estimates the sound pressure p(r,Ω) at the virtual microphone using the sound field coefficient β n,m . Note that a configuration for reducing the computational amount described in Estimation Methods 1 to 5 may be adopted.

[0064] (Estimation Method 7) First, the conditions of Estimation Method 7 will be described. It is necessary to satisfy the conditions of Estimation Method 7 and Estimation Method 5. Further, in Estimation Method 7, the following Condition 10 is satisfied.

[0065] (Condition 10) Embed at least one or more speakers in the rigid sphere B'.

[0066] By using one or more speakers 94-m, ANC by ear sound pressure prediction and local reproduction can be realized with a single device (see Fig. 13). m = 1, 2, …, M, where M is the number of speakers embedded in the rigid sphere B'.

[0067] By providing a speaker inside the rigid sphere B', the requirements of Kirchhoff's integral equation (it is not desired to have a sound source inside the radius to be extrapolated) can be satisfied in deriving the sound pressure prediction formula, and an effect of improving the extrapolation accuracy of the sound pressure can be expected.

[0068] The sound pressure estimation unit 120 uses the observed values of sound pressure [p(a,Ω1), p(a,Ω2),…, p(a,Ω L )] obtained from the output signals (acoustic signals) x(e) of the L error microphones 93-i, and obtains the sound field coefficient β n,m by Equation (9), and estimates the sound pressure p(r,Ω) at the virtual microphone by Equation (8) using the sound field coefficient β n,m . Note that a configuration for reducing the amount of calculation described in Estimation Method 1 may be adopted.

[0069] <Suppression signal generation unit 110> The suppression signal generation unit 110 takes the acoustic signal x(r) and the estimated acoustic signal x(v) as inputs, generates (S110), and outputs a cancellation signal y for suppressing the noise at the installation position of the virtual microphone 130.

[0070] As a method for generating the cancellation signal, a conventional technique can be used. For example, the method of Non-Patent Document 1 can be used. In the present embodiment, a feedforward type ANC is realized by the acoustic signal x(r), the estimated acoustic signal x(v), and the cancellation signal y. The acoustic 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, the noise from the noise source is detected by the reference microphone 91, and the 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 the 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 of the virtual microphone 130 is minimized. Since a conventional update method can be used as the method for updating the coefficients of the noise control filter, the description thereof is 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.

[0071] <Effect> With the above configuration, when it is not possible to place the error microphone near the user's ear, it is possible to achieve high suppression performance without installing a plurality of error microphones to cover the user's head around the user's head.

[0072] <Other Modifications> The present invention is not limited to the above-described embodiments and modifications. For example, the various processes described above may be executed not only in time series according to the description, but also in parallel or individually according to the processing ability of the device that executes the process or as necessary. In addition, various changes can be made as appropriate without departing from the spirit of the present invention.

[0073] <Program and Recording Medium> The various processes described above can be implemented by causing the recording unit 2020 of the computer 2000 shown in FIG. 14 to read a program for executing each step of the above method and causing the control unit 2010, the input unit 2030, the output unit 2040, the display unit 2050, etc. to operate.

[0074] The program describing this processing content can be recorded on a computer-readable recording medium. As the computer-readable recording medium, for example, any of a magnetic recording device, an optical disk, a magneto-optical recording medium, a semiconductor memory, etc. may be used.

[0075] In addition, the distribution of this program is performed, for example, by selling, transferring, lending, etc. a portable recording medium such as a DVD or a CD-ROM on which the program is recorded. Further, the program may be stored in the storage device of a server computer, and the program may be distributed by transferring the program from the server computer to other computers via a network.

[0076] A computer that executes such a program first stores, for example, a program recorded on a portable recording medium or a program transferred from a server computer in its own storage device. Then, when executing the process, the computer reads the program stored in its own recording medium and executes the process according to the read program. As another execution form of this program, the computer may directly read the program from the portable recording medium and execute the process according to the program. Further, each time a program is transferred from the server computer to this computer, the computer may sequentially execute the process according to the received program. Also, the transfer of the program from the server computer to this computer may not be performed, and the above-described process may be executed by a so-called ASP (Application Service Provider) type service that realizes the processing function only by the execution instruction and result acquisition. Note that the program in this embodiment includes information used for processing by an electronic computer and similar to the program (data having a property of defining the processing of the computer but not being a direct instruction to the computer).

[0077] Also, in this embodiment, the present apparatus is configured by causing a computer to execute a predetermined program, but at least a part of these processing contents may be realized hardware-wise.

Claims

1. A generating device that generates a cancellation signal for use in active noise control, generating a cancellation signal such that a point where the amount of noise suppression is maximized is located closer to the user side than the installation positions of a plurality of error microphones, wherein the plurality of error microphones are arranged on a spherical surface close to the head of the user, and the center of the sphere forming the spherical surface is located outside the head of the user, a sound pressure estimation unit that estimates a sound pickup signal picked up when a virtual microphone is installed at a position closer to an observation point than the plurality of error microphones by using spherical harmonic expansion coefficients, and obtains an estimated sound pickup signal x(v); a suppression signal generation unit that generates a cancellation signal for suppressing noise at the installation position of the virtual microphone by using the sound pickup signal x(r) that picks up the noise to be suppressed and the estimated sound pickup signal x(v); wherein the virtual microphone is a microphone that is virtually installed without being actually installed, a generating device.

2. The generating device according to claim 1, wherein 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, a generating device.

3. The generating device according to claim 2, wherein the position of the virtual microphone is the position of the user's ear, and is on the surface of the virtual sphere, a generating device.

4. The generating device according to claim 1, wherein the plurality of error microphones are arranged 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, a generating device.

5. The generating device according to claim 1, wherein the number of the error microphones is 3, the three error microphones are arranged on a spherical surface of a virtual sphere close to the head of the user, 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 expansion coefficients by the least squares method, and obtains the estimated sound pickup signal x(v) by using the estimated spherical harmonic expansion coefficients, a generating device.

6. The generating device according to claim 1, wherein the plurality of error microphones are arranged on a spherical surface of a rigid sphere that reflects sound 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 outside the rigid sphere, a generating device.

7. A generation method for generating a cancellation signal used for active noise control, generating a cancellation signal such that a point where the noise suppression amount is maximized is located closer to the user side than the installation positions of a plurality of error microphones, the plurality of error microphones are arranged on a spherical surface close to the head of the user, and are arranged such that the center of the sphere forming the spherical surface is located outside the head of the user, a sound pressure estimation step of estimating a sound pickup signal picked up when a virtual microphone is installed at a position closer to an observation point than the plurality of error microphones by using spherical harmonic function expansion coefficients, and obtaining an estimated sound pickup signal x(v); a suppression signal generation step of generating a cancellation signal for suppressing noise at the installation position of the virtual microphone by using a sound pickup signal x(r) that picks up noise to be suppressed and the estimated sound pickup signal x(v), the virtual microphone is a microphone that is virtually installed without being actually installed, a generation method.

8. A program for causing a computer to function as the generation device according to any one of Claims 1 to 6.

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