JP2023121933A - Generation device, generation method and program - Google Patents

Abstract

To achieve high suppression performance even at the ear of a user away from an actual error microphone by using a sound collection signal obtained by estimation instead of a sound collection signal collected at an arrangement position of the actual error microphone for estimating the sound collection signal obtained when it is collected at the ear of the user from the sound collection signal collected at the arrangement position of the actual error microphone and actively controlling a cancel signal.SOLUTION: A generation device generates a cancel signal used for active noise control. The generation device generates the cancel signal so that a point in which a suppression amount of noise is maximum is located closer to a user side than an installation position of an error microphone.SELECTED DRAWING: Figure 3

Description

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

Non-Patent Document 1 is known as a conventional active noise suppression technique. Active noise suppression commonly uses reference microphones, error microphones, and canceling speakers. FIG. 1 shows a configuration example of a conventional noise suppression device. A reference microphone 91 picks up the noise emitted by the noise source. The canceling speaker 92 reproduces the canceling signal generated by the suppression signal generating device 90 and emits a canceling sound that cancels out the noise. Furthermore, the error microphone 93 picks up the unerased noise and feeds it back. The suppression signal generation device 90 uses the sound pickup signal of the reference microphone 91 and the sound pickup signal of the error microphone 93 to actively control and generate a cancellation signal so that the amount of unerased noise is small. At the installation position of the error microphone 93, the canceling speaker 92 emits the canceling sound so that the remaining noise is minimized. Therefore, the error microphone 93 is installed near 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 near the user's ear. In some cases, noise is most efficiently suppressed at , and noise remains largely unerased near the user's ear, resulting in a decrease in suppression performance and the user not being able to fully obtain the benefits of noise suppression. For example, if the distance from the noise source to the ear is 100 mm, and the error microphone 93 is placed near the user's ear (0 mm), the suppression performance is -∞ dB, and the error microphone 93 is placed at the midpoint between the noise source and the ear. It was confirmed by simulation that the suppression performance is -7.38dB when installed at . 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 .

The present invention estimates a picked-up signal obtained when a sound is picked up near the user's ear from a picked-up signal picked up at the position where an actual error microphone is arranged, and actively controls a cancellation signal. A generation device that achieves high suppression performance even at the user's ear, which is far from the actual error microphone, by using the sound pickup signal obtained by estimation instead of the sound pickup signal picked up at the position of the error microphone. The object is to provide a generation method and its program.

To solve the above problems, according to one aspect of the present invention, a generation device generates a cancellation signal used for active noise control. The generation device generates a cancellation signal such that the point where the amount of noise suppression is maximized is located closer to the user than the installation position of the error microphone.

ADVANTAGE OF THE INVENTION According to this invention, it is effective in the ability to implement|achieve suppression performance higher than before, when an error microphone cannot be arrange|positioned near a user's ear.

FIG. 3 is a diagram for explaining conventional active noise control; FIG. 4 is a diagram for explaining a suppressible region of the conventional technology; The functional block diagram of the noise suppression system which concerns on 1st embodiment. The figure which shows the example of the processing flow of the noise suppression system which concerns on 1st embodiment. FIG. 4 is a diagram for explaining estimation method 1; FIG. 4 is a diagram for explaining estimation method 2; FIG. 10 is a diagram for explaining estimation method 3; The figure which shows the simulation result of 1st embodiment. The figure which shows the structural example of the computer which applies this method.

Embodiments of the present invention will be described below. In the drawings used for the following description, the same reference numerals are given to components having the same functions and steps performing the same processing, and redundant description will be omitted. In the following explanation, symbols such as ``^'' and `` ^- '' used in the text should be written directly above the characters immediately after them, but due to restrictions in the text notation, they are written just before the characters in question. . These symbols are written in their original positions in the formulas. Further, unless otherwise specified, the processing performed for each element of a vector or matrix is applied to all the elements of the vector or matrix.

<Points of the first embodiment>
In this embodiment, the observed sound pressure at the ear is estimated from the sound signal picked up by an error microphone installed at a position away from the ear. For example, from the sound signal picked up by the actual error microphone, the sound signal picked up by a virtual error microphone placed near the ear is estimated, and in ANC, the sound signal picked up by the virtual error microphone is compared to the sound picked up by the conventional error microphone. used as a signal. 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 that cancels the unerased sound at the ear can be produced.

Various methods are conceivable as a method of estimating the sound signal picked up by the virtual error microphone. For example, the sound pressure is estimated in consideration of the distance attenuation and phase delay from the actual installation position of the error microphone to the ear. Also, for example, from an actual error microphone placed on a spherical surface, the sound pressure around the ear is estimated using spherical harmonics.

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

The noise suppression system includes a reference microphone 91 , a cancellation speaker 92 , an error microphone 93 , a suppression signal generator 110 and a sound pressure estimator 120 . A device comprising the suppression signal generator 110 and the sound pressure estimator 120 is also called a suppression signal generator.

The suppression signal generator receives the sound signal x(r) picked up by the reference microphone 91 and the sound signal x(e) picked up by the error microphone 93. A cancel signal (hereinafter, also referred to as “suppression signal”) y is generated so as to be located closer to the user than the installation position, and output to the cancel speaker 92 .

The suppression signal generator is a special program configured by reading a special program into a known or dedicated computer having, for example, a central processing unit (CPU: Central Processing Unit), a main memory (RAM: Random Access Memory), etc. device. The suppression signal generator executes each process under the control of, for example, a central processing unit. The data input to the suppression signal generator and the data obtained in each process are stored in, for example, a main memory device, and the data stored in the main memory device are read out to the central processing unit as necessary. Used for other processing. 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 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 a suppression signal generation device inside it, and is configured by an auxiliary storage device composed of a semiconductor memory device such as a hard disk, an optical disk, or a flash memory. It may be configured to be provided outside the generation device.

Each part will be described below.

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

<Cancel speaker 92>
The cancel speaker 92 receives the cancel signal y and reproduces the cancel signal y (S92). When the reproduced sound reproduced from the canceling speaker 92 and the noise to be suppressed are in completely opposite phases, the reproduced sound and the noise to be suppressed overlap, that is, the sound waves overlap each other, and the waves cancel each other. noise is suppressed.

<Error Microphone 93>
The error microphone 93 picks up the sound that is not suppressed in the reproduced sound reproduced from the canceling speaker 92, including the unerased noise (S93), and outputs the picked-up sound signal x(e). The error microphone 93 is arranged at a position closer to the noise source than the observation point (for example, near the user's ear). For example, the error microphone 93 is placed 0.05 m closer to the noise source than the ear, as shown in FIG.

<Sound pressure estimation unit 120>
The sound pressure estimator 120 receives the output signal (collected sound signal) x(e) of the error microphone 93, and estimates that the sound is collected when the microphone 130 is installed at a position closer to the observation point than the error microphone 93. Calculate and output the estimated collected sound signal x(v), which is the signal to be obtained. That is, the sound pressure estimating unit 120 estimates a picked-up sound signal obtained when the sound that is not suppressed by the reproduced sound reproduced from the canceling speaker 92 is picked up at the installation position of the microphone 130 (S120), and estimates The resulting collected sound signal is output as an estimated collected sound signal x(v). Three methods of estimating the estimated collected sound signal x(v) are exemplified below. Here, the microphone 130 is not actually installed but installed virtually, and is hereinafter referred to as the virtual microphone 130 .

(Estimation method 1)
In this estimation method, based on the distance attenuation and phase delay from the error microphone 93 and the virtual microphone 130, the sound signal x(v) picked up by the virtual microphone 130 is estimated from the sound signal x(e) actually picked up by the error microphone 93. . FIG. 5 is a diagram for explaining the positional relationship between the noise source, the error microphone 93, and the virtual microphone 130. FIG.

In this estimation method, the position of the noise source is assumed, and noise propagates from the noise source to the error microphone 93 and the virtual microphone 130 as plane waves. By estimating 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 (the position of the virtual microphone 130), the virtual A sound signal picked up by the microphone 130 is estimated. The sound pressure estimator 120 estimates the sound pickup signal of the virtual microphone 130 from the output signal (pickup sound signal) x(e) of the error microphone 93 by the following equation, and the estimated sound pickup signal x(v)=[^ outputs G _p1 ^G _p2 ].

^G _pn =w _n x(e) (n=1,2) (1)
where w _n is
w _n =|G _pn |/|G _e | (2)
and w _n is
w _n =exp((arg G _pn -arg G _e )j) (3)
is. G _e and G _pn in equations (2) and (3) are calculated in advance from the position of the assumed noise source and the observation point prior to the estimation process. For example, place a speaker for the noise source at the assumed position of the noise source, reproduce a predetermined signal with the speaker for the noise source, and obtain _Ge from the sound signal picked up by the microphone placed at the position of the error microphone. G _pn is obtained from the collected sound signal picked up by the microphone placed at the position of the observation point.
(Estimation method 2)
In this estimation method, the collected sound signals of a virtual error microphone are estimated from the collected sound signals of a plurality of 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 actual error microphones.

In this estimation method, error microphones are placed at regular intervals on a spherical surface of radius r _e to estimate the sound pressure on the spherical surface of radius r. For example, the distance from the center to the error microphone is r _e =0.15 m, and (i) six error microphones are arranged at the center of each face of the regular hexahedron (see (i) in FIG. 6), (ii) 12 By arranging the error microphones at the center of each face of the regular dodecahedron (see (ii) in FIG. 6), the error microphones can be arranged at equal intervals. For example, the distance from the center to the observation point (the position of the virtual microphone 130) is estimated as r=0.08m.

By using the spherical harmonic 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.

Sound _pressure observation values p(θ ₁ , φ ₁ ) _, p(θ ₂ , φ _{2 )} , . For example, let x(e)=[p(θ ₁ , φ ₁ ), p(θ ₂ , φ ₂ ), . . . , p(θ _L , φ _L )].

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 estimator 120 obtains the sound field coefficient P _nm (r _e ) on the radius r _e for the spherical harmonic function Y ^m _n (·) using the following equation.

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

where k is the wave number, a is the radius of the sound-reflecting rigid sphere, j _n is the spherical Bessel function of order n, j _n ' is the derivative of j _n , and h _n ⁽²⁾ is the second is the seed sphere Hankel function, and h _n ' ⁽²⁾ is the derivative of h _n ⁽²⁾ .
The sound pressure estimator 120 obtains an estimated sound pressure value ^p(r, θ, φ) at the observation point (r, θ, φ) through resynthesis.

Assume that the estimated collected sound signal x(v)=^p(r, θ, φ).

The derivation of Equation (5) will be described 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。 Assuming that the noise source is a point sound source and considering the reflection from a rigid sphere with a radius of a, the sound pressure at the point (r, θ, φ) is

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

By transforming equation (10) into B _nm =... and substituting it into (9),

becomes.

Note that the maximum order N in the spherical harmonic expansion is subject to the following restrictions.

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

Furthermore, N is subject to the following constraints as a condition that spatial aliasing does not occur.

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 estimable region is limited to within the distance r=0.18 m from the head.

(Estimation method 3)
In this estimation method, from the collected sound signals of multiple error microphones placed at non-equidistant intervals near the head, the spherical harmonic expansion coefficients estimated by the least-squares method are used to estimate the collected sound signals of a virtual error microphone. to estimate FIG. 7 is a diagram for explaining the positional relationship of error microphones. For example, 4 points behind the head (4 azimuth angles (0°, 30°, 150°, 180°) x 0° elevation angle) or 12 points behind the head (4 azimuth angles (0°, 30°, 150°) , 180°) × three elevation angles (-30°, 0°, 30°)). The installation radius of the error microphone may be determined according to the installation environment and the type of sound to be suppressed. For example, if you want to suppress running noise, consider the seat size and set the error microphone installation radius to 0.13 m.

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

なお、ω_i＝(θ_i,φ_i)とし、i=1,2,…,Lとする。 Sound pressure observation values p(r _e ,θ ₁ ,φ ₁ ),p(r _e ,θ ₂ , _{φ 2} ),…,p(r _e ,θ _L , φ _L ). Sound pickup signal 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 represented as follows.

Note that ω _{i =} (θ _i , φ _i ) and 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 estimator 120 obtains ^−P (r _e ) that minimizes the squared error of the absolute value as a solution.

Assuming that the point to be estimated is on the radius r, calculation of the expansion coefficient P _nm (r) and the estimated value are the same as in estimation method 2. In other words,
The sound pressure estimator 120 obtains the sound field coefficient P _nm (r _e ) on the radius r _e for the spherical harmonic function Y ^m _n (·) using the following equation.

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

The sound pressure estimator 120 obtains an estimated sound pressure value ^p(r, θ, φ) at the observation point (r, θ, φ) through resynthesis.

Assume that the estimated collected sound signal x(v)=^p(r, θ, φ).

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

A conventional technique can be used as a method of generating the cancel signal. For example, the method of 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. estimating a sound pickup signal that would be obtained when the virtual microphone 130 detects interference sound between the noise from the noise source and the reproduced sound of the cancellation signal y, and detecting the noise from the noise source with the reference microphone 91; A cancel signal y is generated by inputting it to a noise control filter realized by an adaptive digital filter, and reproduced by a cancel 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 adaptive algorithm updates the coefficients of the noise control filter so that the input of 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, the description is omitted. In feedforward ANC, a secondary path model estimating the secondary path is used to compensate the effect of the secondary path in the adaptive algorithm.

<effect>
With the above configuration, it is possible to achieve higher suppression performance than in the past when the error microphone cannot be placed near the user's ear. FIG. 8 shows simulation results of the first embodiment. In (A) the noise is a plane wave of 300Hz, and in (B) the noise is a plane wave of 100Hz.

<Other Modifications>
The present invention is not limited to the above embodiments and modifications. For example, the various types of processing described above may not only be executed in chronological order according to the description, but may also be executed in parallel or individually according to the processing capacity of the device that executes the processing or as necessary. In addition, appropriate modifications are possible without departing from the gist of the present invention.

<Program and recording medium>
The various processes described above can be performed by loading a program for executing each step of the above method into the storage unit 2020 of the computer shown in FIG. .

A program describing the contents of this processing can be recorded in a computer-readable recording medium. Any computer-readable recording medium may be used, for example, a magnetic recording device, an optical disk, a magneto-optical recording medium, a semiconductor memory, or the like.

Also, the distribution of this program is carried out by selling, assigning, lending, etc. portable recording media such as DVDs and CD-ROMs on which the program is recorded. Further, the program may be distributed by storing the program in the storage device of the server computer and transferring the program from the server computer to other computers via the network.

A computer that executes such a program, for example, first stores the program recorded on a portable recording medium or the program transferred from the server computer once in its own storage device. Then, when executing the process, this computer reads the program stored in its own recording medium and executes the process according to the read program. Also, as another execution form of this program, the computer may read the program directly from a portable recording medium and execute processing according to the program, and the program is transferred from the server computer to this computer. Each time, the processing according to the received program may be executed sequentially. In addition, the above processing is executed by a so-called ASP (Application Service Provider) type service, which does not transfer the program from the server computer to this computer, and realizes the processing function only by the execution instruction and result acquisition. may be It should be noted that the program in this embodiment includes information that is used for processing by a computer and that conforms to the program (data that is not a direct instruction to the computer but has the property of prescribing the processing of the computer, etc.).

Moreover, in this embodiment, the device is configured by executing a predetermined program on a computer, but at least a part of these processing contents may be implemented by hardware.

Claims (8)

A generator for generating a cancellation signal used for active noise control,
Generating a cancellation signal so that the point where the amount of noise suppression is maximized is located closer to the user than the installation position of the error microphone.
generator. The generator of claim 1, comprising:
a sound pressure estimator for estimating a sound pickup signal picked up when a virtual microphone is installed at a position closer to an observation point than the error microphone, and obtaining an estimated sound pickup signal x(v);
A suppression signal for generating a cancellation signal for suppressing noise at the installation position of the virtual microphone using the collected sound signal x(r) obtained by collecting the noise to be suppressed and the estimated collected sound signal x(v). a generator;
The virtual microphone is a microphone that is virtually installed without actually being installed.
generator. The generator of claim 2, wherein
The sound pressure estimator obtains the estimated collected sound signal x(v) from the collected sound signal x(e) based on distance attenuation and phase delay from the positional relationship between the error microphone and the virtual microphone.
generator. The generator of claim 2, wherein
The sound pressure estimating unit uses spherical harmonic expansion coefficients from the collected sound signals x(e) of the plurality of error microphones placed near the user's head at equal intervals to obtain the estimated collected sound signal get x(v),
generator. The generator of claim 2, wherein
The sound pressure estimating unit uses spherical harmonic expansion coefficients estimated by the least squares method from the collected sound signals x(e) of the plurality of error microphones arranged at non-equidistant intervals near the user's head. to obtain the estimated sound pickup signal x(v),
generator. A generation method for generating a cancellation signal used for active noise control, comprising:
Generating a cancellation signal so that the point where the amount of noise suppression is maximized is located closer to the user than the installation position of the error microphone.
generation method. The method of claim 6, comprising:
a sound pressure estimation step of estimating a sound pickup signal picked up when the virtual microphone is 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 for generating a cancellation signal for suppressing noise at the installation position of the virtual microphone using the collected sound signal x(r) obtained by collecting the noise to be suppressed and the estimated collected sound signal x(v). a generating step;
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 the generation device according to any one of claims 1 to 5.

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