Sound field control apparatus and method for controlling sound field
원문보기
IPC분류정보
국가/구분
United States(US) Patent
등록
국제특허분류(IPC7판)
H04R-029/00
H04R-003/00
H04S-007/00
출원번호
US-0080310
(2011-04-05)
등록번호
US-9002019
(2015-04-07)
우선권정보
JP-2010-091818 (2010-04-12)
발명자
/ 주소
Ise, Tomohiko
출원인 / 주소
Alpine Electronics, Inc.
대리인 / 주소
Brinks Gilson & Lione
인용정보
피인용 횟수 :
2인용 특허 :
4
초록▼
A sound field control apparatus includes at least two main microphones; for each main microphone, a set of at least two sub microphones arranged such that the at least two sub microphones are placed in different axis directions about each of the main microphones; a filtering unit; and a filter coeff
A sound field control apparatus includes at least two main microphones; for each main microphone, a set of at least two sub microphones arranged such that the at least two sub microphones are placed in different axis directions about each of the main microphones; a filtering unit; and a filter coefficient calculating unit configured to calculate a filter coefficient for the filtering unit. A filter coefficient used to control sound pressure levels and air particle velocities of an output audio signal is calculated on the basis of a sound pressure level detected by each main microphone and the difference between the sound pressure level detected by the main microphone and that detected by each of the corresponding sub microphones.
대표청구항▼
1. A sound field control apparatus comprising: at least two main microphones arranged at points of measurement in a space;for each main microphone of the at least two main microphones, at least two sub microphones associated with the main microphone arranged such that the at least two sub microphone
1. A sound field control apparatus comprising: at least two main microphones arranged at points of measurement in a space;for each main microphone of the at least two main microphones, at least two sub microphones associated with the main microphone arranged such that the at least two sub microphones are placed in different axis directions about the main microphone that the at least two sub microphones are associated with, where each of the sub microphones is not a main microphone;a filtering unit configured to filter an input audio signal;at least one speaker configured to output the audio signal filtered by the filtering unit; anda filter coefficient calculating unit in communication with the filtering unit, the filter coefficient calculating unit configured to calculate a filter coefficient, used to control sound pressure levels and air particle velocities of the audio signal output from the speaker in the space, for the filtering unit on the basis of a sound pressure level detected by each main microphone and the difference between the sound pressure level detected by the main microphone and that detected by each of the corresponding sub microphones;wherein the filter coefficient calculating unit is configured to obtain an acoustic system transfer function of sound pressure level on the basis of a sound pressure level detected by each main microphone, to obtain a sound pressure gradient by dividing a difference between the sound pressure level detected by the main microphone and that detected by each of the corresponding sub microphones by a distance between the main microphone and the sub microphone, to convert the sound pressure gradients into air particle velocities to obtain acoustic system transfer functions of air particle velocity, and to calculate the filter coefficient on the basis of the acoustic system transfer function of sound pressure level and the acoustic system transfer functions of air particle velocity. 2. The apparatus according to claim 1, wherein the sound field control apparatus comprises three sub microphones associated with each main microphone and wherein the filter coefficient calculating unit is configured to calculate the air particle velocities using the following expression: vx1(x,ω)=1jωρ0p(x1,x2,x3,ω)-p(x1+Δx1,x2,x3,ω)Δx1vx2(x,ω)=1jωρ0p(x1,x2,x3,ω)-p(x1,x2+Δx2,x3,ω)Δx2vx3(x,ω)=1jωρ0p(x1,x2,x3,ω)-p(x1,x2,x3+Δx3,ω)Δx3 where vx1, vx2, and vx3 denote air particle velocities in the x1-axis, x2-axis, and x3-axis directions, p denotes the sound pressure level, and p0 denotes the density of air. 3. The apparatus according to claim 2, wherein the filter coefficient calculating unit is configured to calculate the filter coefficient using the following expression: w(ω)=[C(ω) Bx1(ω) Bx2(ω) Bx3(ω)]T+ h(ω) where w denotes the filter coefficient, C denotes the acoustic system transfer function of sound pressure level, Bx1, Bx2, and Bx3 denote the acoustic system transfer functions of air particle velocity in the x1-axis, x2-axis, and x3-axis directions, and h denotes a target transfer function of air particle velocity. 4. The apparatus according to claim 2, wherein the filter coefficient calculating unit is configured to calculate the filter coefficient using the following expression: w(ω)=[αpC(ω) αvx1Bx1(ω) αvx2Bx2(ω) αvx3Bx3(ω)]T+ h(ω) where w denotes the filter coefficient, C denotes the acoustic system transfer function of sound pressure level, Bx1, Bx2, and Bx3 denote the acoustic system transfer functions of air particle velocity in the x1-axis, x2-axis, and x3-axis directions, h denotes a target transfer function of air particle velocity, and αp, αvx1, αvx2, and αvx3 denote weighting factors. 5. The apparatus according to claim 2, wherein the filter coefficient calculating unit is configured to calculate the filter coefficient on the basis of an LMS algorithm with an adaptive filter using the following expression: w(n+1,ω)=w(n,ω)+2μu*(ω)[C(ω) Bx1(ω) Bx2(ω) Bx3(ω)]TH E(ω) where w denotes the filter coefficient, C denotes the acoustic system transfer function of sound pressure level, Bx1, Bx2, and Bx3 denote the acoustic system transfer functions of air particle velocity in the x1-axis, x2-axis, and x3-axis directions, μ denotes a step size parameter, n denotes the number of sequential computation updates by the adaptive filter, u* denotes the conjugate complex number of the input audio signal u, and E denotes an error. 6. The apparatus according to claim 2, wherein the filter coefficient calculating unit is configured to calculate the filter coefficient on the basis of an LMS algorithm with an adaptive filer using the following expression: w(n+1,ω)=w(n,ω)+2μu*(ω)[αpC(ω) αvx1Bx1(ω) αvx2Bx2(ω) αvx3Bx3(ω)]TH E(ω) where w denotes the filter coefficient, C denotes the acoustic system transfer function of sound pressure level, Bx1, Bx2, and Bx3 denote the acoustic system transfer functions of air particle velocity in the x1-axis, x2-axis, and x3-axis directions, μ denotes a step size parameter, n denotes the number of sequential computation updates by the adaptive filter, u* denotes the conjugate complex number of the input audio signal u, E denotes an error, and αp, αvx1, αvx2, and αvx3 denote weighting factors. 7. A computer-implemented method for controlling a sound field in an acoustic system including a filtering unit configured to filter an input audio signal and at least one speaker configured to output the audio signal filtered by the filtering unit, the method comprising: calculating a filter coefficient used to control sound pressure levels and air particle velocities of the audio signal output from the at least one speaker in the space on the basis of a sound pressured level detected by each of at least two main microphones arranged at points of measurement in a space and a difference between the sound pressure level detected by a main microphone of the at least two main microphones and that detected by each set of sub microphones associated with the main microphone, each set of sub microphones comprising at least two sub microphones placed in different axis directions about each main microphone of the at least two main microphones, where each of the sub microphones is not a main microphone,setting the calculated filter coefficient in the filtering unit;wherein calculating the filter coefficient comprises: obtaining an acoustic system transfer function of sound pressure level on the basis of a sound pressure level detected by each main microphone,obtaining a sound pressure gradient by dividing a difference between the sound pressure level detected by the main microphone and that detected by each of the corresponding sub microphones by a distance between the main microphone and the sub microphone,converting the sound pressure gradients into air particle velocities to obtain acoustic system transfer functions of air particle velocity, andcalculating the filter coefficient on the basis of the acoustic system transfer function of sound pressure level and the acoustic system transfer functions of air particle velocity.
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이 특허에 인용된 특허 (4)
Adkins Charles N. (Falls Church VA) Turtora John J. (Fairfax VA), Adaptive signal processing array with unconstrained pole-zero rejection of coherent and non-coherent interfering signals.
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