JPH11168792A - Sound field controller - Google Patents

Abstract

PROBLEM TO BE SOLVED: To provide a sound field controller, capable of correcting transmission characteristics over an entire acoustic space by few loudspeakers and an adaptive filters. SOLUTION: This sound field controller is provided with control filters 102 which include an adaptive filter, M pieces of loudspeakers 104, K pieces of microphones 106, a mode division filter 108 for leading out N' pieces of made amplitudes from respective sound pressures (p) of the microphones 106, N' pieces of arithmetic parts 110 for calculating errors of the respective mode amplitudes against the mode amplitude as a target, N' pieces of mode area error weighting parts 112 for weighting the errors of respective modes and an area conversion filter 114 for converting the error of a mode area into the error of a time area. By the control filter 102, an adaptive processing is performed based on the error to a target response of the mode amplitudes of the respective modes.

Description

DETAILED DESCRIPTION OF THE INVENTION

[0001]

BACKGROUND OF THE INVENTION 1. Field of the Invention The present invention relates to a sound field control device for controlling a transfer characteristic of a sound field to be uniform regardless of a position.

[0002]

2. Description of the Related Art Generally, in an acoustic space, reflected waves and standing waves are generated by walls and the like, and sound waves mutually interfere with each other, so that acoustic transfer characteristics are complicatedly disturbed. In particular, in a narrow space such as a vehicle interior where sound such as glass is easily reflected, reflected waves and standing waves have a large effect. Is big. An adaptive equalization system is known as a technique for correcting such disturbance of the sound transfer characteristic. According to the adaptive equalization system, a predetermined sound field space can be realized at an arbitrary control point.

FIG. 14 is a diagram showing a configuration of an adaptive equalization system applied to an audio device. The adaptive equalization system shown in the figure includes an audio source 500, a target response setting unit 501, a microphone 502, a calculation unit 504, an adaptive signal processing device 506, and a speaker 508. The audio source 500 includes a radio tuner, a CD player, and the like, and outputs an audio signal x (n). The target response setting section 501 has a target response characteristic (impulse response) H set therein, receives an audio signal x (n) output from the audio source 500, and receives a corresponding target response signal d (n). Is output. The microphone 502 is installed at a listening position (control point) in the vehicle interior acoustic space, detects a sound at this observation point, and outputs a music signal d ′ (n). Arithmetic unit 50
4 is a music signal d ′ output from the microphone 502
An error between (n) and the target response signal d (n) output from the target response setting unit 501 is calculated to output an error signal e (n). The adaptive signal processing device 506 generates the error signal e (n)
Generates a signal y (n) so that the power of the signal y becomes minimum.
The speaker 508 emits a sound corresponding to the signal y (n) output from the adaptive signal processing device 506 to the vehicle interior acoustic space.

The target response characteristic H of the target response setting section 501
Has a characteristic corresponding to the sound field space to be reproduced. For example, assuming that a delay time of about half of the number of taps of the adaptive filter is t, the delay time t has the flat characteristic (gain 1 characteristic) in the entire audio frequency band.
Is set. The delay time t is used by the adaptive filter to accurately approximate the inverse characteristic of the acoustic system, and the target response setting unit 5 having such a target response characteristic.
01 can be realized by setting the tap coefficient corresponding to the delay time t of the FIR (Finite Impulse Response) type digital filter to 1, and setting the other tap coefficients to 0.

The adaptive signal processing device 506 receives the audio signal x (n) as a reference signal, the error signal e (n) output from the arithmetic unit 504, and the error signal e (n). Adaptive signal processing is performed so that the power of n) is minimized, and a signal y (n) is output.
The adaptive signal processing device 506 is a LMS (Least Mean Square).
e) The algorithm processing unit 510, the adaptive filter 512 having a FIR digital filter configuration, and the audio signal x (n) adapted by convolving the propagation characteristic (transfer characteristic) C of the sound propagation system from the speaker 508 to the listening position. Reference signal (filtered reference signal) u used for signal processing
(N).

[0006] The LMS algorithm processing unit 510 includes an error signal e (n) at the listening position and a signal processing filter 51.
4 is input, and the LMS algorithm is used so that the music signal d '(n) at the listening position becomes equal to the target response signal d (n) using these signals. Is used to set the tap coefficient vector W of the adaptive filter 512. Adaptive filter 512
Performs digital filter processing on the audio signal x (n) using the tap coefficient vector W set as described above, and outputs a signal y (n).

The error signal e is obtained by such adaptive processing.
Adaptive filter 512 so that the power of (n) is minimized.
When the tap coefficient vector W converges, it is possible to listen to the same music as when listening to music in a space having the target response characteristic H set in the target response setting unit 501.

[0008]

By the way, in the above-mentioned adaptive equalization system, it is possible to listen to music with a transfer characteristic similar to the target response characteristic H at the control point.
No guarantees are made for the characteristics other than the control points. For this reason, if an attempt is made to listen to ideal music at many positions in the acoustic space by the adaptive equalization system, a large number of control points must be set, and a correspondingly large number of speakers are required. In addition, installing a large number of speakers as control sound sources means that the number of adaptive filters 512 required for that is also increased, which leads to an increase in the circuit scale and the amount of calculation.

SUMMARY OF THE INVENTION The present invention has been made in view of the above circumstances, and has as its object to provide a sound field control device capable of correcting the transfer characteristics over the entire acoustic space with a small number of speakers and adaptive filters. It is in.

[0010]

In order to solve the above-mentioned problems, a sound field control device according to the present invention is provided with a plurality of speakers and a plurality of microphones at predetermined positions in an acoustic space. The sound pressure distribution is mode-decomposed based on the output signal, and the control is performed so that the mode amplitude of each mode becomes a predetermined value. By controlling the mode amplitude of each mode, it is possible to reduce or cancel the influence of the mode in which the sound pressure changes greatly when the listening position is moved. Therefore, in particular, increase the number of control points (listening positions). Instead, the transfer characteristics can be corrected over the entire acoustic space with a small number of speakers and adaptive filters, and a flat sound pressure distribution can be realized.

In order to control the mode amplitude of each mode described above, an adaptive process is performed on a signal in a time domain,
A case is considered where adaptive processing is performed on a signal in the mode region. In any case, the mode amplitude of each mode can be controlled at the installation position of the microphone, and the transfer characteristics can be corrected over the entire acoustic space.

In particular, when performing processing on a signal in the mode domain, by performing adaptive processing including processing for returning to a signal in the time domain before being input to the speaker, an inverse filter of the acoustic system can be calculated. This eliminates the need for a process of deriving a signal to be actually input to the speaker using the obtained signal, thereby simplifying the process.

Preferably, the plurality of speakers are arranged at positions other than the positions corresponding to the nodes of the vibration of the mode to be controlled. Although the mode amplitude of the corresponding mode cannot be controlled by locating the speaker at the position of the node of vibration, the mode amplitude of the mode can be reduced or canceled by arranging it at a position where it is removed. Various controls can be performed.

Further, it is preferable that each of the plurality of speakers outputs the input signal with the same sign, and is arranged at all positions where the sign of the vibration of the mode to be canceled is opposite. As a result, it is possible to cancel only other desired modes while leaving the zero-order mode.

The above-described control for each mode may not be performed for all the modes, but may be performed only for a part of the modes, preferably, for a low-order mode excluding the zero-order mode. In general, since the mode amplitude of a low-order mode such as the first-order or second-order mode is large, by controlling only this low-order mode, the transfer characteristics can be efficiently corrected over the entire acoustic space with a small amount of calculation. .

[0016]

DETAILED DESCRIPTION OF THE PREFERRED EMBODIMENTS A sound field control apparatus according to an embodiment of the present invention performs mode decomposition of a sound pressure distribution and controls the sound pressure level of each of the decomposed modes so that the entire sound space can be controlled. It is characterized by correcting the transfer characteristics. Hereinafter, a sound field control device according to an embodiment will be described with reference to the drawings.

(1) Mode decomposition In order to control the modes in the acoustic space, it is necessary to perform mode decomposition of the sound pressure distribution. The procedure of the mode decomposition is shown below. A wave equation of a one-dimensional sound field having M sound sources therein and having both ends closed is given by the following equation (1). The one-dimensional sound field refers to a sound field in which the sound pressure changes only in a predetermined axial direction x.

[0018]

(Equation 1)

Here, x is the position of the microphone, ω is the angular frequency, p (x, ω) is the sound pressure, q _m is the input signal to the m-th speaker, and l _m is the m-th speaker. , N is the number of all speakers, ξ _{n '} is the attenuation ratio on the wall of the n'th mode, N' is the number of all modes, L is the length of the sound field,
ω _{n ′} (= n′πc ₀ / L) is the natural frequency of the sound field, ρ
₀ is the air density, c ₀ is the speed of sound, δ (n ′) is n ′ = 0
And Kronecker's delta function which becomes 1 when n ′ ≠ 0 and 1 when n ′ ≠ 0.

In the equation (1),

[0021]

(Equation 2)

[0022]

(Equation 3)

Where one of a _{n ′} (ω) is the amplitude of the n ′ mode and ψ _{n ′} (x) is the eigenfunction of the n ′ mode.

In the above equation (1), p (x, ω)
Is the sound pressure at a distance x of the microphone in the one-dimensional sound field, so that K points (x ₁ ,
The sound pressure p (x, ω) at each microphone when a microphone is installed at (x ₂ ,..., x _K ) is represented by the following matrix notation.

[0025]

(Equation 4)

Here,

[0027]

(Equation 5)

Is as follows. When the equation (4) is rewritten using the mode eigenfunction Ψ,

[0029]

(Equation 6)

## EQU1 ##

The following equation (7) is obtained by multiplying both sides of the equation (6) by the inverse matrix (inverse mode eigenfunction) ^{-1 -1} of the eigen matrix (mode eigenfunction) from the left.

[0032]

(Equation 7)

From equation (7), the sound pressure p at each microphone
The amplitude a _{n ′} (ω) of each mode can be obtained from (x _K , ω). According to the above procedure, the mode decomposition of the sound pressure distribution is performed.

FIG. 1 is a diagram showing a specific example of a mode decomposition section configured by applying the mode decomposition method. The mode decomposition unit 10 shown in FIG. 1 includes M speakers 2, K microphones 4, and a mode decomposition filter 6 that derives N mode amplitudes from the sound pressures of the microphones 4. When the signals q _{1 to} q _M are input to the _M speakers 2 and the sound is emitted to the one-dimensional sound field of the acoustic system C, the sound pressures p _{1 to} p _{K at the} microphones 4 are respectively (4) Given by the formula. The mode resolution filter 6 controls these sound pressures p _{1 to} p _K
Is input, and from the mode 0 to the mode N−
The mode amplitudes a _{0 to} a _N−1 of ₁ are calculated and output.

FIG. 2 is a diagram showing frequency characteristics of each mode included in the acoustic system. As shown in FIG.
There are modes of each order such as next, second, ...
By performing control to reduce or cancel a mode in which the sound pressure level greatly fluctuates when the listening position of the audio sound is moved, it is possible to obtain a transmission characteristic with little fluctuation over the entire acoustic space. .

FIGS. 3A and 3B show the amplitude states of the modes. FIG. 3A shows the amplitude state of the zero-order mode, and FIG. 3B shows the amplitude state of the first-order mode. . FIG.
As shown in (a), in the 0th-order mode, the entire sound space vibrates in the same phase, so that the audio sound can be heard at the same sound pressure level regardless of the listening position. However, as shown in FIG. 3B, in the primary mode, the sound pressure level greatly varies depending on the listening position. Therefore, when the first-order mode component in the sound radiated into the acoustic space is large, the first-order mode component is reduced or negated so that the acoustic characteristics are substantially uniform even when the listening position is moved. A sound field can be realized. The same applies to each of the second and higher modes. If the second or higher order component is large, control is performed to reduce or cancel the component.

(2) Mode Control Next, the mode amplitude obtained by the mode decomposition is converted to an LMS
A sound field control device that performs control using an algorithm will be described. The LMS algorithm includes an algorithm in which an adaptive filter operates in the time domain and an LMS algorithm in which the adaptive filter operates in the mode domain. The purpose of controlling the mode amplitude is the same, but they have different system configurations. Hereinafter, the sound field control devices of the first to third embodiments having three types of algorithms will be described.

(2-1) Sound Field Controller of First Embodiment Having Algorithm for Operating Adaptive Filter in Time Domain The sound field controller of the first embodiment is controlled by an LMS algorithm operating in the time domain. The adaptive filter has an adaptive filter, and the error calculated in the mode domain is re-converted to the time domain to update the coefficient of the adaptive filter.

FIG. 4 is a diagram showing a schematic configuration of the sound field control device according to the first embodiment. As shown in the figure, the sound field control device of the present embodiment includes a control filter 102 including M adaptive filters with the number of taps of I, and M speakers 10.
4, K microphones 106, and microphones 10
6, a mode division filter 108 as mode decomposition means for deriving N ′ mode amplitudes from each sound pressure p, and N ′ operation units 110 for calculating an error of each mode amplitude with respect to a target mode amplitude; The apparatus includes N ′ mode region error weighting units 112 for weighting errors in each mode, and a region conversion filter 114 for converting a mode region error into a time domain error.

The output signal y _m (n) of the m-th control filter 102 is input signal u (n) and the control filter 10
The convolution with the coefficient w _{m of} 2 is expressed by the following equation (8).

[0041]

(Equation 8)

The output signal y _m (n) is input to the m-th speaker 104, a sound is radiated to a one-dimensional sound field of the acoustic system C, and is taken into each microphone 106. The sound pressure p _k (n) at the k-th microphone 106 is given by the following equation.

[0043]

(Equation 9)

Here, c _km (j) is the m-th speaker 1
The j-th tap coefficient of the acoustic system C from 04 to the k-th microphone 106 is shown, and w _m (i) is the i-th tap coefficient of the m-th control filter 102.
Rewriting equation (9) in matrix expression,

[0045]

(Equation 10)

Is as follows. In equations (9) and (10),

[0047]

[Equation 11]

(Equation 12)

[0048]

(Equation 13)

[0049]

[Equation 14]

[0050]

(Equation 15)

[0051]

(Equation 16)

[0052]

[Equation 17]

Is as follows.

The mode amplitude a (n) can be obtained by performing a mode decomposition on the sound pressure p (n) of the microphone 106 obtained by the equation (10) in the same manner as in the equation (7). it can. That is, the mode division filter 108

[0055]

(Equation 18)

The mode amplitude a
(N) is derived. In equation (18),

[0057]

[Equation 19]

Is as follows.

On the other hand, the output d _k (n) of the k-th target impulse response output from the target response setting section (described later)
Is given by the following equation (20).

[0060]

(Equation 20)

Here, h _k (s) indicates the coefficient of the s-th tap of the k-th target impulse response. Rewriting equation (20) in matrix expression,

[0062]

(Equation 21)

Is obtained. In equations (20) and (21),

[0064]

(Equation 22)

[0065]

(Equation 23)

[0066]

(Equation 24)

[0067]

(Equation 25)

Is as follows. Mode amplitude d '(n) of target response
Can be obtained by performing mode decomposition on the target response signal d _k (n) obtained by Expression (21) by the same method as Expression (7). Therefore, the mode amplitude d ′ (n) of the target response is

[0069]

(Equation 26)

Is given by In equation (26),

[0071]

[Equation 27]

Is as follows.

The error e ′ (n) in the mode area is
The mode amplitude d ′ of the target response given by equation (26)
Mode amplitude a (n) given from (n) to (18)
Can be obtained by subtracting Therefore,
The calculation unit 110

[0074]

[Equation 28]

An error e ′ (n) in the mode region is derived by the operation given by In equation (28),

[0076]

(Equation 29)

Is as follows.

Next, the mode area error weighting section 112
The error e mode region to select the mode of controlling _{'(n) (e' 0} (n) ~e ' against _N-1 (n)), the weighting coefficient B (b ₀ ~b _{N' -1} ). The domain conversion filter 114 calculates the time domain error e (n) by multiplying the weighted mode domain error by the mode eigenfunction Ψ. Mode region error e '
The weighting for (n) and the conversion from the weighted mode domain error to the time domain error are:

[0079]

[Equation 30]

Is given by In equation (30),

[0081]

(Equation 31)

[0082]

(Equation 32)

Is as follows.

Here, the error vector e in the time domain
When the instantaneous estimated value of the gradient vector of the error characteristic surface is obtained by partially differentiating the instantaneous power e (n) ^T e (n) of (n) with the filter coefficient w,

[0085]

[Equation 33]

[0086] Therefore, the control filter 10
The update of the coefficient of 2 is performed by the following equation.

[0087]

(Equation 34)

Here, μ is a step size parameter (coefficient for controlling the magnitude of correction in each repetition) of the LMS algorithm.

Next, a detailed configuration of the sound field control device according to the first embodiment will be described. FIG. 5 is a diagram illustrating an overall configuration of the sound field control device according to the first embodiment. As shown in the figure, the sound field control device 100 includes a control filter 102 including M adaptive filters with the number of taps I, and M speakers 1
04, K microphones 106, mode division filters 108, N 'operation units 110, N' mode region error weighting units 112, region conversion filters 114, target response setting units 116, mode division filters 118, filtered An x unit 120 and an LMS algorithm processing unit 122 are provided.

The control filter 102, the speaker 104,
The microphone 106, the mode division filter 108, the calculation unit 110, the mode region error weighting unit 112, and the region conversion filter 114 perform the operations described with reference to FIG.

The target response setting section 116 sets a characteristic (target response characteristic H) corresponding to the sound field space to be reproduced, for example, a characteristic having a delay time of about half the number of taps of the filter constituting the control filter 102. ing. The mode division filter 118 derives N ′ mode amplitudes from the target response signal output from the target response setting unit 116,
Output to the arithmetic unit 110.

Filtered x section 120 receives input signal u
This is a filter for creating a reference signal from (n).
Specifically, the filtered x section 120 performs the above-described C
It is configured by connecting filters having the characteristics of ＾, Ψ ⁻¹ , B, and Ψ in series. LMS algorithm processing unit 122
Configures the control filter 102 based on the time domain error signal e (n) output from the domain conversion filter 114 and the reference signal output from the filtered x section 120 in accordance with the above equation (34). Adjust the filter coefficients of the adaptive filter.

As described above, the sound pressure distribution is mode-decomposed, and
By controlling a mode having a large amplitude, that is, a mode that adversely affects the transfer characteristics of the acoustic space, the transfer characteristics of the entire acoustic space can be corrected.

Next, a modification of the sound field control device of the first embodiment will be described. As shown in FIG. 2, normally, except for the 0th order, the mode amplitude increases as the order decreases. Therefore, by controlling only the lower order modes,
Almost desired acoustic characteristics can be realized, and the amount of processing can be reduced. However, as can be seen from FIG. 2, since the high-order mode signal excluded from the control target contains many high-frequency components, it is preferable to eliminate the high-order mode signal itself because the high-frequency component is reduced. Absent. Thus, at least one of the M speakers 104 also functions as an uncontrolled sound source so that the input signal u
It is preferable to input (n) itself.

FIG. 6 is a diagram showing a modification of the sound field control device of the first embodiment, and shows the configuration of a sound field control device that performs control only in a low-order mode. The sound field control device 150 shown in the figure outputs the input signal u (n) directly from the speaker 104 without passing through the control filter 102. In order to adjust the delay amount at that time, the delay device 152 Equipped. In the delay unit 152, a delay time obtained by subtracting the delay time when the signal passes through the acoustic system C from the delay time set in the target response setting unit 116 is set as the delay amount β. Also, the mode region error weighting unit 112
For example, only the weighting coefficient bm of the mode to be controlled becomes 1,
Others are set to 0, and only the error signal of the mode to be controlled is input to the domain conversion filter 114, and the control filter 102 controls only some of the modes.

As described above, control is performed only in some modes, and in other modes, an input signal is output from the speaker 104 as it is, thereby realizing a sound field in which sound pressure fluctuation due to movement of the listening position is small. And the amount of calculation can be reduced.

(2-2) The sound field control device of the second embodiment having an algorithm for operating the adaptive filter in the mode domain The above-described sound field control device of the first embodiment operates the adaptive filter in the time domain. Had an algorithm to
The operation may be performed according to an algorithm for operating the adaptive filter in the mode region. In order to operate in the mode domain, the error calculated in the mode domain may be used as it is for updating the coefficients of the adaptive filter.

FIG. 7 is a diagram showing a schematic configuration of a sound field control device according to the second embodiment. As shown in the figure, the sound field control device of the present embodiment includes an acoustic system modeling filter 202 simulating the acoustic system C, and an acoustic system modeling filter 20.
A mode division filter 204 that derives N ′ mode amplitudes from a signal (sound pressure) output from the control signal 2, a control filter 206 including N ′ adaptive filters with the number of taps I,
A domain conversion filter 208 that converts a mode-domain signal output from the control filter 206 into a time-domain signal.
An acoustic system inverse filter 210 that restores the acoustic system C 模 simulated by the acoustic system modeling filter 202;
Speakers 212, K microphones 214,
A mode division filter 216 for deriving N ′ mode amplitudes from the sound pressure of the microphone 214, N ′ operation units 218 for calculating errors in each mode, and N ′ modes for weighting errors in each mode Area error weighting section 220
And

When the adaptive filter is operated in the mode domain, the coefficients of the control filter 206 are obtained in the mode domain. Therefore, the input signal to the control filter 206 must be a signal in the mode domain. For this reason, once
The input signal u (n) is passed through an acoustic modeling filter 202 having the same characteristics as the actual acoustic system C, and then the time-domain signal output from the acoustic modeling filter 202 is output by the mode division filter 204 to the mode domain. Is converted to a signal.

When actually outputting sound from the speaker 212, the signal input to the speaker 212 must be a signal in the time domain. For this reason, the domain conversion filter 208 converts the mode-domain signal output from the control filter 206 into a time-domain signal again. The signal in the time domain output from the domain conversion filter 208 is the signal (microphone 21) after passing through the acoustic system C ＾ by the acoustic system modeling filter 202.
4), the signal is returned to a signal corresponding to the position of the speaker 212 by passing the signal through the acoustic inverse filter 210.

Incidentally, the k-th output signal p of the acoustic system modeling filter 202 that models the acoustic system C
_k (n) is a convolution of the input signal u (n) and the acoustic system modeling filter 202,

[0102]

(Equation 35)

Is represented by When equation (35) is rewritten in matrix expression,

[0104]

[Equation 36]

The following is obtained. These equations (35) and (36)
In the formula:

[0106]

(37)

(38)

[0107]

[Equation 39]

[0108]

(Equation 40)

[0109]

[Equation 41]

Is as follows.

The mode amplitude a ＾ of the output of the modeling filter
(N) can be obtained by multiplying the output signal p ＾ (n) of the acoustic system modeling filter 202 obtained by the equation (36) by the inverse mode eigenfunction Ψ ⁻¹ . Therefore, the mode division filter 204

[0112]

(Equation 42)

The mode amplitude a ＾
(N) is derived. In equation (42),

[0114]

[Equation 43]

Is as follows. This mode amplitude a ＾ (n) becomes an input signal of the control filter 206. Therefore, the output signal y (n) of the control filter 206 is

[0116]

[Equation 44]

Is obtained. In equation (44),

[0118]

[Equation 45]

[0119]

[Equation 46]

Is as follows. Equation (44) is

[0121]

[Equation 47]

It can be rewritten as follows. (4
In equation 7),

[0123]

[Equation 48]

[0124]

[Equation 49]

[0125]

[Equation 50]

Is as follows. ψ _n'-1 ^-1 is the inverse mode eigenfunction Ψ
^{-1 is} a vector composed of elements in the n'th row.

Next, the area conversion filter 208 outputs the output signal y of the control filter 206 which is a signal in the mode area.
(N) is multiplied by a mode eigenfunction Ψ to convert it into a signal in the time domain. Further, since the signal in the time domain is a signal simulated by the acoustic system modeling filter 202 into the acoustic system C ＾, the acoustic system inverse filter 210 applies the inverse filter F of the acoustic system C ＾ to restore the original signal. . Therefore, the output signal y ′ (n) of the acoustic inverse filter 210 is

[0128]

(Equation 51)

Is as follows. here,

[0130]

(Equation 52)

[0131]

(Equation 53)

Is as follows.

This output signal y ′ (n) is transmitted to the speaker 212
And the sound is radiated to the one-dimensional sound field of the acoustic system C, and is captured by the microphone 214. The sound pressure p (n) at the microphone 214 is

[0134]

(Equation 54)

Is given by here,

[0136]

[Equation 55]

[0137]

[Equation 56]

[0138]

[Equation 57]

Is as follows.

The mode amplitude a (n) can be obtained by performing mode decomposition on the sound pressure p (n) at the microphone 214 obtained by the equation (54) in the same manner as in the equation (7). it can. Therefore, the mode division filter 2
16 is

[0141]

[Equation 58]

The mode amplitude a
(N) is derived. here,

[0143]

[Equation 59]

Is as follows.

On the other hand, the mode amplitude d '(n) of the target response
Is similar to equation (26).

[0146]

[Equation 60]

Is given by here,

[0148]

[Equation 61]

(Equation 62)

[0149]

[Equation 63]

[0150]

[Equation 64]

Is as follows.

The error e ′ (n) in the mode area is
Mode amplitude d 'of the target response given by equation (60)
Mode amplitude a (n) given from (n) to (58)
Can be obtained by subtracting Therefore,
The calculation unit 218

[0153]

[Equation 65]

The error given in the mode area e ′ (n) is calculated by performing the calculation given by here,

[0155]

[Equation 66]

Is as follows.

Next, the mode area error weighting section 220
Performs weighting of the error e ′ (n) in the mode area by the weighting coefficient B according to the following equation (67).

[0158]

[Equation 67]

Here,

[0160]

[Equation 68]

[0161]

[Equation 69]

Is as follows.

When the instantaneous power e (n) ^T e (n) of the weighted error vector e (n) in the mode region is partially differentiated with the filter coefficient w, an instantaneous estimated value of the gradient vector of the error characteristic surface is obtained.

[0164]

[Equation 70]

Is obtained. Therefore, the control filter 20
The coefficient 6 is updated by the following equation.

[0166]

[Equation 71]

Here, μ is a step size parameter of the LMS algorithm, and is a coefficient for controlling the magnitude of correction in each repetition.

Next, a detailed configuration of the sound field control device according to the second embodiment will be described. FIG. 8 is a diagram illustrating an overall configuration of a sound field control device according to the second embodiment. As shown in the drawing, the sound field control device 200 includes an acoustic system modeling filter 202, a mode division filter 204, and N ′ of the number of taps I.
Filter 206 including the number of adaptive filters, area conversion filter 208, acoustic system inverse filter 210, M speakers 212, K microphones 214, mode division filter 216, N 'arithmetic units 218, N' Mode region error weighting section 220, target response setting section 222,
A mode division filter 224, a filtered x section 226,
An LMS algorithm processing unit 228 is provided.

Acoustic system modeling filter 202, mode division filter 204, control filter 206, area conversion filter 208, acoustic system inverse filter 210, speaker 2
12, microphone 214, mode division filter 21
6, arithmetic unit 218, mode region error weighting unit 220
Perform the operations described with reference to FIG.

The target response setting section 222 has a characteristic (target response characteristic H) corresponding to the sound field space to be reproduced, for example, a characteristic having a delay time of about half the number of taps of the filter constituting the acoustic inverse filter 210. Is set. The mode division filter 224 derives N ′ mode amplitudes from the target response signal output from the target response setting section 222 and outputs the mode amplitudes to the calculation section 218.

The filtered x section 226 is a filter for creating a reference signal from the mode amplitude a ＾ (n) which is the output signal of the mode division filter 204. Specifically, the filtered x section 226 performs the above {, C},
It is configured by connecting filters having the characteristics of F, Ψ ⁻¹ , and B in series. The LMS algorithm processing unit 228 includes:
Mode region error signal e (n) output from mode region error weighting section 220 and filtered x section 226
(71) based on the reference signal output from
The filter coefficient of the adaptive filter constituting the control filter 206 is adjusted according to the equation.

As described above, by performing control by the control filter 206 in the mode region, it is possible to control a mode having a large amplitude, that is, a mode having a bad influence on the transfer characteristics of the acoustic space, and the transfer characteristics of the entire acoustic space can be controlled. Can be corrected.

Next, a modification of the sound field control device according to the second embodiment will be described. The sound field control device of the present embodiment includes:
As in the first embodiment in which control is performed in the time domain, only some of the modes (particularly, the lower-order mode) may be controlled.

FIG. 9 is a diagram showing a modification of the sound field control device according to the second embodiment, and shows the configuration of a sound field control device which controls only a part of the modes. The sound field control device 250 shown in FIG.
2, the input signal u (n) is output directly from the speaker 212, bypassing the acoustic inverse filter 210.
A delay unit 252 is provided to adjust the amount of delay at that time. The delay unit 252 includes a target response setting unit 222
The delay time obtained by subtracting the delay time of the acoustic system C from the delay time set in (1) is set as the delay amount β. Further, in the control filter 206, for example, a tap coefficient corresponding to a higher-order mode that is not a control target is set to 0, and filter operation and control using the operation result are performed only in some modes. ing.

FIG. 10 is a diagram showing a modification of the sound field control device shown in FIG. Sound field control device 2 shown in FIG.
A similar purpose of controlling only a part of the modes can be achieved by passing the uncontrolled mode components through the amplifier 262 having a coefficient of 1 without passing through the control filter 206 as in the case of 60. .

(2-3) Sound Field Control Device of Third Embodiment Having Algorithm for Operating Adaptive Filter in Mode Domain The sound field control device of the second embodiment described above includes an acoustic inverse filter 210. There is a need. Therefore, the preparation procedure of calculating the acoustic inverse filter 210 when configuring the system requires one more step. Further, an error may occur in the inverse filter F of C ＾ depending on the fluctuation of the acoustic system C, and accurate control may not be performed. Therefore,
In the sound field control device according to the third embodiment, the area conversion filter 208 and the acoustic inverse filter 210 used in the sound field control device according to the second embodiment are incorporated in the control filter 206.

FIG. 11 is a diagram showing a schematic configuration of a sound field control device according to the third embodiment. As shown in the figure, the sound field control device of the present embodiment includes an acoustic system modeling filter 302 simulating the acoustic system C and an acoustic system modeling filter 3.
A mode division filter 304 for deriving N ′ mode amplitudes from the signal (sound pressure) output from the control filter 02 and a control filter 3 including N ′ × M adaptive filters with I taps
06, M speakers 312, K microphones 314, a mode division filter 316 for deriving N 'mode amplitudes from the sound pressure of the microphones 314, and N' operations for calculating errors in each mode A unit 318 and N ′ mode region error weighting units 320 for weighting errors in each mode.

The control filter 306 included in the sound field control device of this embodiment is the same as the control filter 2 shown in FIG.
06, the domain conversion filter 208 and the acoustic inverse filter 21
0 is taken in. Therefore, when the output signal y ′ (n) of the acoustic inverse filter 210 shown in FIG.

[0179]

[Equation 72]

Is obtained. Here, FΨW is reset to W again, that is, the control filter 206 shown in FIG.
Assuming that the control filter 306 is obtained by incorporating the domain conversion filter 208 and the acoustic inverse filter 210 into the control signal 306, the output signal y ′ (n) of the control filter 306 is

[0181]

[Equation 73]

Is given by In equation (73),

[0183]

[Equation 74]

[0184]

[Equation 75]

[0185]

[Equation 76]

[0186]

[Equation 77]

[0187]

[Equation 78]

[0188]

[Expression 79]

[0189]

[Equation 80]

[0190]

[Equation 81]

[0191]

(Equation 82)

[0192]

[Equation 83]

Is as follows.

This output signal y ′ (n) is supplied to the speaker 312
And the sound is emitted to the one-dimensional sound field of the acoustic system C, and is taken in by the microphone 314. The sound pressure p (n) at the microphone 314 is

[0195]

[Equation 84]

Is given by here,

[0197]

[Equation 85]

[0198]

[Equation 86]

[0199]

[Equation 87]

Is as follows.

The mode amplitude a (n) is calculated based on the sound pressure p (n) of the microphone 314 obtained by the equation (84).
It can be obtained by performing mode decomposition in the same manner as in equation (7). Therefore, the mode division filter 3
16 is

[0202]

[Equation 88]

The mode amplitude a
(N) is derived. here,

[0204]

[Equation 89]

Is as follows.

On the other hand, the mode amplitude d '(n) of the target response
Is similar to equation (26).

[0207]

[Equation 90]

Is given by here,

[0209]

[Equation 91]

[0210]

(Equation 92)

[0211]

[Equation 93]

[0212]

[Equation 94]

[0213]

An error e ′ (n) in the mode area is
Mode amplitude d 'of the target response given by equation (90)
Mode amplitude a (n) given from (n) to (88)
Can be obtained by subtracting Therefore, the arithmetic unit 318 calculates

[0215]

[Equation 95]

The error given in the mode area e ′ (n) is calculated by performing the calculation given by here,

[0219]

[Equation 96]

[0219]

Next, mode area error weighting section 320
Performs weighting of the error e ′ (n) in the mode region with the weighting coefficient B. This weight is

[0220]

(97)

Is calculated by In equation (97),

[0222]

[Equation 98]

[0223]

[Equation 99]

Is as follows.

Here, the instantaneous power e (n) ^T e (n) of the weighted error vector e (n) in the mode region is partially differentiated with the filter coefficient w, thereby obtaining the instantaneous estimated value of the gradient vector of the error characteristic surface. When asked,

[0226]

[Equation 100]

Is obtained. Therefore, the control filter 30
The coefficient 6 is updated by the following equation.

[0228]

[Equation 101]

Here, μ is a step size parameter of the LMS algorithm, and is a coefficient for controlling the magnitude of correction in each repetition.

Next, a detailed configuration of the sound field control device according to the third embodiment will be described. FIG. 12 is a diagram illustrating a detailed configuration of the sound field control device according to the third embodiment. As shown in the figure, the sound field control device 300 includes a control filter 30 including an acoustic system modeling filter 302, a mode division filter 304, and N ′ × M adaptive filters with the number of taps I.
6, M speakers 312, K microphones 31
4. Mode division filter 316, N 'operation units 31
8, N ′ mode region error weighting units 320, target response setting units 322, mode division filters 324, filtered x units 326, and LMS algorithm processing units 328.

Acoustic system modeling filter 302, mode division filter 304, control filter 306, speaker 312, microphone 314, mode division filter 31
6, arithmetic unit 318, mode region error weighting unit 320
Perform the operations described with reference to FIG.

The target response setting section 322 sets a characteristic (target response characteristic H) corresponding to the sound field space to be reproduced, for example, a characteristic having a delay time of about half the number of taps of the filter constituting the control filter 306. Have been. The mode division filter 324 derives N ′ mode amplitudes from the target response signal output from the target response setting unit 322, and outputs the N ′ mode amplitudes to the calculation unit 318.

The filtered x section 326 is a filter for creating a reference signal from the mode amplitude a ＾ (n) which is the output signal of the mode division filter 304. Specifically, the filtered x section 326 is configured by connecting filters having respective characteristics of C ＾, Ψ ⁻¹ , and B in series.
The LMS algorithm processing unit 328 outputs a mode region error signal e output from the mode region error weighting unit 320.
Based on (n) and the reference signal output from the filtered x section 326, the filter coefficient of the adaptive filter forming the control filter 306 is adjusted in accordance with the above-described equation (101).

As described above, the control filter 30 incorporating the area conversion filter 208 and the acoustic inverse filter 210
By using 6, it is not necessary to obtain the inverse filter F in advance, so that the system preparation procedure can be reduced by one step. Further, since the inverse filter F is incorporated in the control filter, it is possible to cope with some fluctuations in the acoustic system C.

Next, a modification of the sound field control device according to the third embodiment will be described. The sound field control device of the present embodiment includes:
As in the first and second embodiments described above, only some of the modes (particularly, the lower-order modes) may be controlled.

FIG. 13 is a diagram showing a modification of the sound field control device according to the third embodiment, and shows the configuration of a sound field control device that controls only a part of the modes. The sound field control device 350 shown in FIG.
02, the input signal u (n) is directly output from the speaker 312 by bypassing from the control filter 306,
A delay unit 352 is provided to adjust the amount of delay at that time. The delay unit 352 includes a target response setting unit 322
The delay time obtained by subtracting the delay time of the acoustic system C from the delay time set in (1) is set as the delay amount β. In the control filter 306, for example, tap coefficients corresponding to higher-order modes not to be controlled are set to 0, so that filter operation and control using the operation result are performed only in some modes. Has become.

(3) Number of Speakers and Arrangement Method Next, the number of speakers included in the sound field control device of each of the above-described embodiments and an optimal arrangement method will be described. Equation (2) above is the mode amplitude of each mode when the sound pressure distribution is mode-decomposed. The variable that can be controlled in the equation (2) is q that is the input signal to the m-th speaker.
_m (ω) and l _m which is the position of the m-th speaker
One. Therefore, for example, when canceling the n'th mode, that is, when setting the mode amplitude of the n'th mode to 0,

[0238]

[Equation 102]

Must be satisfied.

However, in equation (102), cos
If term (n'πl _m / L) is to place the speaker becomes zero such a position, the mode control very according to the speaker what value input signal q _k (omega) is the speaker becomes impossible Therefore, the speaker cannot be arranged at such a position. Also, if the speaker is arranged at a position where cos (n'πl _m / L) becomes as close to 1 as possible, the input signal q _k (ω) greatly affects the control of the mode amplitude, that is, the input to the control Signal q
_Since the efficiency of _k (ω) increases, it is desirable to arrange the speaker at such a position.

As shown in FIG. 3A, the mode shape of the zero-order mode is flat, and as shown in FIG. 3B, the mode shape of the first-order mode (the same applies to other modes). The mode shape has a peak dip. Therefore, as described above, in order to realize a flat sound pressure distribution in the entire acoustic space, it is necessary to cancel the other modes while leaving the zero-order mode.

In order to find the mode amplitude of the zero-order mode,
If n ′ = 0 in the equation (2),

[0243]

[Equation 103]

Is obtained. This equation (103) is
Since the term of (n′πl _m / L) is not included, the mode amplitude of the 0th-order mode depends only on the sum of the input signals q _k (ω) to the respective speakers regardless of the position of the speakers. It is shown that. That is, if the input signals q _k (ω) to the respective speakers have different values, the signals cancel each other out when summing up, and the amplitude of the zero-order mode becomes small. Therefore, the input signal q _k (ω) to each speaker must always have the same sign. In order to cancel another mode under this condition, cos (n′πl _m /
It is necessary to arrange speakers at all positions where L) has the opposite sign.

It should be noted that the present invention is not limited to the above embodiment, and various modifications can be made within the scope of the present invention. For example, in the above-described embodiment, the mode control in the case of a one-dimensional sound field has been described, but the case of a three-dimensional sound field can be similarly considered. The wave equation in the case of a three-dimensional sound field uses the following equation (104) instead of the above-described equation (1).

[0246]

[Equation 104]

Note that x ₁ , x ₂ , and x ₃ represent the vertical, horizontal, and height positions of the microphone, ω represents the angular frequency, and p (x ₁ , x
₂ , x ₃ , ω) are the sound pressures, q _m is the input signal to the m-th speaker, l _1m , l _2m , l _3m are the vertical, horizontal and height positions of the m-th speaker, M Is the total number of speakers, ξ _n'1 ,
ξ _n'2 , ξ _n'3 are the attenuation ratios on the wall surfaces of the n ' ₁ , n' ₂ , and n ' ₃ modes, N' is the total number of modes, L ₁ , L ₂ , L
₃ is the length, width and height of the sound field, ω _{n'1, n'2, n'3} (=
πc ₀ ｛(n ' ₁ / L ₁ ) ² + (n ′ ₂ / L ₂ ) ² +
(N ' ₃ / L ₃ ) ² ｝) is the characteristic frequency of the sound field, ρ ₀
Indicates the air density, and c ₀ indicates the speed of sound.

In the equation (104),

[0249]

[Equation 105]

[0250]

[Equation 106]

Is as follows.

[0252]

As described above, according to the present invention, by controlling the mode amplitude of each mode of the sound field, the influence of the mode in which the sound pressure changes greatly when the listening position moves is reduced. Therefore, the transfer characteristics can be corrected over the entire acoustic space with a small number of speakers and adaptive filters without increasing the number of control points, and a flat sound pressure distribution can be realized.

In this case, by arranging the plurality of speakers at positions other than the positions corresponding to the nodes of the vibration of the mode to be controlled, various types of control such as reducing the mode amplitude of the mode or canceling the mode are performed. Becomes possible. In addition, each of the plurality of speakers outputs the input signal with the same sign, and is preferably arranged at all positions where the sign of the vibration of the mode to be canceled is opposite,
As a result, it is possible to cancel only other desired modes while leaving the zero-order mode.

[Brief description of the drawings]

FIG. 1 is a diagram illustrating a specific example of a mode decomposition unit configured by applying a mode decomposition method.

FIG. 2 is a diagram illustrating frequency characteristics of each mode included in the acoustic system.

FIG. 3 is a diagram illustrating an amplitude state of a mode.

FIG. 4 is a diagram illustrating a schematic configuration of a sound field control device according to the first embodiment.

FIG. 5 is a diagram illustrating an overall configuration of a sound field control device according to the first embodiment.

FIG. 6 is a diagram illustrating a modified example of the sound field control device of the first embodiment.

FIG. 7 is a diagram illustrating a schematic configuration of a sound field control device according to a second embodiment.

FIG. 8 is a diagram illustrating an overall configuration of a sound field control device according to a second embodiment.

FIG. 9 is a diagram illustrating a modified example of the sound field control device of the second embodiment.

FIG. 10 is a diagram showing a modification of the sound field control device shown in FIG.

FIG. 11 is a diagram illustrating a schematic configuration of a sound field control device according to a third embodiment.

FIG. 12 is a diagram illustrating a detailed configuration of a sound field control device according to a third embodiment.

FIG. 13 is a diagram illustrating a modified example of the sound field control device of the third embodiment.

FIG. 14 is a diagram illustrating a configuration of an adaptive equalization system applied to an audio device.

[Explanation of symbols]

 REFERENCE SIGNS LIST 100 sound field control device 102 control filter 104 speaker 106 microphone 108, 118 mode division filter 110 operation unit 112 mode region error weighting unit 114 region conversion filter 116 target response setting unit 120 filtered x unit 122 LMS algorithm processing unit

Claims (9)

[Claims] 1. A sound field control device, wherein a sound pressure distribution in an acoustic space is mode-decomposed and controlled for each mode. 2. A plurality of speakers installed at a predetermined position in an acoustic space and emitting an input signal into the acoustic space; and a plurality of speakers installed in the acoustic space and emitted from the plurality of speakers. A plurality of microphones for collecting sound; a mode decomposition means for modulating a sound pressure distribution based on output signals of the plurality of microphones; and a mode amplitude of each mode decomposed by the mode decomposition means to a predetermined value. And a control filter for controlling the input signals input to the plurality of speakers. 3. The sound field control device according to claim 2, wherein the control filter controls a mode amplitude of each mode by performing an adaptive process on a signal in a time domain. 4. The control filter according to claim 2, wherein the control filter performs an adaptive process on a signal in a mode region obtained by modulating the input signal.
A sound field control device for controlling a mode amplitude of each mode. 5. The method according to claim 4, wherein the control filter includes a process of performing a process on a signal in a mode region and then returning the signal to a signal in a time domain before being input to the speaker. A sound field control device that performs an adaptive process including the above process. 6. The sound field control device according to claim 2, wherein the plurality of speakers are arranged at positions other than positions corresponding to nodes of vibration of a mode to be controlled. . 7. The speaker according to claim 2, wherein each of the plurality of speakers outputs the input signal with the same sign, and the sign of the vibration of the mode to be canceled is opposite. A sound field control device, wherein the sound field control device is disposed at a position of a sound field. 8. The sound field control device according to claim 1, wherein a part of a mode included in the acoustic space is a control target. 9. The sound field control device according to claim 8, wherein some modes to be controlled are low-order modes other than the zero-order mode.