Patent Translate Powered by EPO and Google Notice This translation is machine-generated. It cannot be guaranteed that it is intelligible, accurate, complete, reliable or fit for specific purposes. Critical decisions, such as commercially relevant or financial decisions, should not be based on machine-translation output. DESCRIPTION JPH11168792 [0001] BACKGROUND OF THE INVENTION 1. Field of the Invention The present invention relates to a sound field control apparatus for controlling transfer characteristics of a sound field to be uniform regardless of position. [0002] 2. Description of the Related Art Generally, in an acoustic space, a reflected wave, a standing wave or the like is generated by a wall or the like, and sound waves interfere with each other to disturb the acoustic transfer characteristic in a complicated manner. In particular, in a narrow space such as a passenger compartment enclosed with a glass-like sound that is likely to be reflected, the influence of the disturbance of the acoustic transfer characteristic on the listening of the sound is significant because the influence of the reflected wave and the standing wave is large. Is big. An adaptive equalization system is known as a technique for correcting such disturbance of the acoustic transfer characteristic. According to the adaptive equalization system, a predetermined sound field space can be realized at any control point. [0003] FIG. 14 is a diagram showing the configuration of an adaptive equalization system applied to an audio device. The adaptive equalization system shown in the figure includes an audio source 500, 08-05-2019 1 a target response setting unit 501, a microphone 502, an arithmetic 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 unit 501 is set with a target response characteristic (impulse response) H, receives the audio signal x (n) output from the audio source 500, and corresponds to the target response signal d (n). Output The microphone 502 is installed at a listening position (control point) in the vehicle interior sound space, detects a sound at this observation point, and outputs a music signal d '(n). The computing unit 504 computes an error between the music signal d '(n) output from the microphone 502 and the target response signal d (n) output from the target response setting unit 501, and outputs an error signal e (n). Do. Adaptive signal processor 506 generates signal y (n) such that the power of error signal e (n) is minimized. The speaker 508 emits a sound corresponding to the signal y (n) output from the adaptive signal processing device 506 into the in-vehicle acoustic space. [0004] The target response characteristic H of the target response setting unit 501 is set to a characteristic corresponding to the sound field space to be reproduced. For example, when a delay time about half the number of taps of the adaptive filter is t, a flat characteristic (characteristic of gain 1) is set in all audio frequency bands with this delay time t. The delay time t is for the adaptive filter to accurately approximate the inverse characteristic of the acoustic system. The target response setting unit 501 having such a target response characteristic is an FIR (Finite Impulse Response) digital This can be realized by setting the tap coefficient corresponding to the delay time t of the filter to 1 and setting the other tap coefficients to 0. [0005] The adaptive signal processing device 506 receives the audio signal x (n) as a reference signal and receives the error signal e (n) output from the above-described operation unit 504, and the adaptive signal processing device 506 receives the error signal e (n). Adaptive signal processing is performed so that the power is minimized, and a signal y (n) is output. The adaptive signal processing unit 506 includes an LMS (Least Mean Square) algorithm processing unit 510, an adaptive filter 512 of an FIR type digital filter configuration, and propagation of an acoustic propagation system from the speaker 508 to the listening position in the audio signal x (n). And a signal processing filter 514 that generates a reference signal (filtered reference signal) u (n) used for adaptive signal processing by convoluting a characteristic (transfer characteristic) C. 08-05-2019 2 [0006] The LMS algorithm processing unit 510 receives the error signal e (n) at the listening position and the reference signal u (n) output from the signal processing filter 514, and uses these signals to use the music signal d at the listening position. The tap coefficient vector W of the adaptive filter 512 is set using the LMS algorithm such that '(n) is equal to the target response signal d (n). The adaptive filter 512 subjects the audio signal x (n) to digital filtering using the tap coefficient vector W set in this manner, and outputs a signal y (n). [0007] If the tap coefficient vector W of the adaptive filter 512 converges such that the power of the error signal e (n) is minimized by such an adaptive process, music is generated in the space having the target response characteristic H set in the target response setting unit 501. It becomes possible to listen to the same music as listening to music. [0008] By the way, although the above-mentioned adaptive equalization system can listen to music with the same transfer characteristic as the target response characteristic H at the control point, the characteristics other than the control point Does not guarantee at all. For this reason, when trying to listen to ideal music at many positions in the acoustic space by the adaptive equalization system, many control points are set, and correspondingly many speakers are required. Further, installing a large number of loudspeakers as control sound sources means that the number of adaptive filters 512 required for that purpose also increases, which leads to an increase in circuit size and computation amount. [0009] The present invention has been made in view of such points, and an object thereof is to provide a sound field control device capable of correcting the transfer characteristic over the entire sound space by using a small number of speakers and an adaptive filter. . [0010] 08-05-2019 3 SUMMARY OF THE INVENTION In order to solve the above-mentioned problems, in the sound field control apparatus of the present invention, a plurality of speakers and a plurality of microphones are installed at predetermined positions in an acoustic space, and each microphone The sound pressure distribution is mode-decomposed based on the output signal of and the mode amplitude of each mode is controlled to be a predetermined value. By controlling the mode amplitude of each mode, it is possible to reduce or cancel out the effects of modes that cause a large change in sound pressure when the listening position moves, and in particular to increase the control point (listening position). Instead, the transmission characteristics can be corrected over the entire acoustic space by a small number of speakers and an adaptive filter to realize a flat sound pressure distribution. [0011] In order to control the mode amplitude of each mode described above, the case of performing adaptive processing on the signal of the time domain and the case of performing adaptive processing on the signal of the mode domain can be considered. In any case, the mode amplitude of each mode can be controlled at the installation position of the microphone, and the transfer characteristic can be corrected over the entire acoustic space. [0012] In particular, when performing processing on a signal in the mode region, the inverse filter of the acoustic system is calculated by performing adaptive processing including processing to restore the signal in the time domain before being input to the speaker. This eliminates the need for processing for deriving a signal actually input to the speaker using the above-mentioned method, and the processing can be simplified. [0013] Further, it is preferable that the plurality of speakers be disposed at positions other than the positions corresponding to the nodes of vibration of the mode to be controlled. Even if a speaker is placed at the position of the vibration node, the mode amplitude of the 08-05-2019 4 corresponding mode can not be controlled, but by placing it at a position where it is removed, the mode amplitude of that mode is reduced or canceled, etc. Control of the [0014] Further, it is preferable that each of the plurality of speakers be arranged at all positions where the signs of the vibration of the mode to be canceled are opposite as well as the signs of the input signal are aligned and output. In this way, it is possible to cancel only the other desired modes while leaving the zero-order mode. [0015] Further, the control for each mode described above may not be performed for all modes, but may be performed for only a partial mode, preferably a low-order mode other than the zero-order mode. Generally, since the mode amplitude of the low-order mode such as first-order and secondorder modes is large, the transfer characteristic can be efficiently corrected over the entire acoustic space with a small amount of calculation by controlling only this low-order mode . [0016] BEST MODE FOR CARRYING OUT THE INVENTION The sound field control device according to one embodiment to which the present invention is applied, performs mode decomposition of sound pressure distribution and controls the sound pressure level of each decomposed mode, thereby covering the entire acoustic space. It is characterized in that the transfer characteristic is corrected. Hereinafter, a sound field control device according to an embodiment will be described with reference to the drawings. [0017] (1) Mode Decomposition In order to control the mode of acoustic space, it is necessary to perform mode decomposition of sound pressure distribution. The procedure of mode decomposition is shown below. The wave equation of a one-dimensional sound field closed at both ends with M sound sources inside is given by the following equation (1). The one- 08-05-2019 5 dimensional sound field refers to a sound field in which the sound pressure changes in accordance with only the predetermined axial direction x. [0018] Here, x is the position of the microphone, ω is the angular frequency, p (x, ω) is the sound pressure, qm is the input signal to the m-th speaker, and lm is the position of the m-th speaker , M is the total number of speakers, ξ n 'is the attenuation ratio at the wall surface of the n'th mode, N' is the total number of modes, L is the length of the sound field, ω n '(= n' π c 0 / L ) Is the characteristic frequency of the sound field, ρ0 is the air density, c0 is the speed of sound, δ (n ') is the 1 function when n' = 0, 1's with n '≠ 0, and the Kronekka delta function is 0 It shows. [0019] Also, in equation (1), [0020] One an '(ω) is the amplitude of the n'th mode, and' n '(x) is the eigenfunction of the n'th mode. [0021] In the above equation (1), p (x, ω) is the sound pressure at the distance x of the microphone in the one-dimensional sound field, so for K points (x1, x2, ..., xK) in the one-dimensional sound field The sound pressure p (x, ω) at each microphone when the microphones are installed is represented by the following matrix notation. [0022] 【００２６】ここで、 [0023] 【００２８】である。 If equation (4) is rewritten using mode eigenfunction Ψ, 08-05-2019 6 [0024] 【００３０】となる。 [0025] By applying the inverse matrix (inverse mode eigenfunction) Ψ−1 of the eigen matrix (mode eigenfunction) to both sides of the equation (6) from the left, the following equation (7) is obtained. [0026] From the sound pressure p (xK, ω) at each microphone, the amplitude an '(ω) of each mode can be obtained from the equation (7). Mode decomposition of sound pressure distribution is performed by the above procedure. [0027] FIG. 1 is a diagram illustrating a specific example of a mode decomposition unit configured by applying a mode decomposition method. The mode decomposition unit 10 shown in the figure includes M speakers 2, K microphones 4, and a mode decomposition filter 6 for deriving N mode amplitudes from the sound pressure of the microphone 4. The sound pressures p1 to pK at the respective microphones 4 when the signals q1 to qM are input to the M speakers 2 and the sound is emitted to the one-dimensional sound field of the acoustic system C are respectively given by the equation (4) . The mode decomposition filter 6 receives these sound pressures p1 to pK, and calculates and outputs mode amplitudes a0 to aN-1 of mode 0 to mode N-1 by the equation (7). 08-05-2019 7 [0028] FIG. 2 is a diagram showing frequency characteristics of each mode included in the acoustic system. As shown in the figure, there are modes of each order, such as 0th order, 1st order, 2nd order,... Modes in which the sound pressure level largely fluctuates when the listening position of the audio voice is moved. By making the control smaller or cancel out, it is possible to obtain a transfer characteristic with less variation over the entire acoustic space. [0029] FIG. 3 is a diagram showing an amplitude state of the mode, and FIG. 3A shows an amplitude state of the zero-order mode, and FIG. 3B shows an amplitude state of the first-order mode. As shown in FIG. 3A, in the zero-order mode, since the vibration is performed in the same phase in the entire acoustic space, it becomes possible to listen to the audio voice 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 largely fluctuates depending on the listening position. Therefore, when the primary mode component in the sound radiated to the acoustic space is large, the acoustic characteristics are substantially uniform even when the listening position is moved by reducing or canceling this. A sound field can be realized. The same applies to the second and higher order modes, and when the second order and higher order components are large, control is performed to reduce or cancel the components. [0030] (2) Mode Control Next, a sound field control apparatus for controlling the mode amplitude obtained by mode decomposition using the LMS algorithm will be described. The LMS algorithm includes one in which the adaptive filter operates in the time domain and one in which the adaptive filter operates in the mode domain. The purpose of controlling the mode amplitude is the same but the system configuration is different. The sound field control devices according to the first to third embodiments having three kinds of algorithms will be described below. 08-05-2019 8 [0031] (2-1) Sound Field Control Device of the First Embodiment Having an Algorithm for Operating an Adaptive Filter in the Time Domain The sound field control device of the first embodiment has an adaptive system controlled by the LMS algorithm operating in the time domain. A filter is provided, and the error calculated in the mode domain is reconverted to the time domain to perform coefficient updating of the adaptive filter. [0032] 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, in the sound field control apparatus of this embodiment, a control filter 102 including M adaptive filters with a tap number of I, M speakers 104, K microphones 106, and a microphone A mode division filter 108 as mode decomposing means for deriving N 'number of mode amplitudes from each sound pressure p of 106, N' number of operation units 110 for calculating an error of each mode amplitude with respect to a target mode amplitude, An N ′ mode region error weighting unit 112 that weights errors in each mode, and a region conversion filter 114 that converts errors in the mode region into errors in the time region are provided. [0033] The output signal ym (n) of the mth control filter 102 is expressed as the following equation (8) as a convolution of the input signal u (n) and the coefficient wm of the control filter 102. [0034] The output signal ym (n) is input to the m-th speaker 104, and the sound is emitted to the onedimensional sound field of the acoustic system C and is taken into each microphone 106. The sound pressure pk (n) at the k-th microphone 106 is given by the following equation. 08-05-2019 9 [0035] Here, ckm (j) is the coefficient of the j-th tap of the acoustic system C from the m-th speaker 104 to the k-th microphone 106, and w m (i) is the i-tap of the m-th control filter 102 The coefficients of the eyes are shown respectively. Rewriting equation (9) in matrix form, [0036] 【００４６】となる。 In equations (9) and (10), [0037] 【００５３】である。 [0038] The mode amplitude a (n) can be obtained by performing mode decomposition on the sound pressure p (n) in the microphone 106 obtained by the equation (10) in the same manner as the equation (7). That is, the mode division filter 108 [0039] The mode amplitude a (n) is derived by the operation given by In equation (18), [0040] 【００５８】である。 [0041] On the other hand, the output dk (n) of the k-th target impulse response output from the target 08-05-2019 10 response setting unit (described later) is given by the following equation (20). [0042] Here, hk (s) indicates the coefficient of the s-th tap of the k-th target impulse response. Rewriting equation (20) in matrix form, [0043] 【００６３】となる。 In equations (20) and (21), [0044] 【００６８】である。 The mode amplitude d ′ (n) of the target response can be obtained by performing mode decomposition on the target response signal dk (n) obtained by equation (21) in the same manner as equation (7). Thus, the mode amplitude d '(n) of the target response is [0045] Given by In equation (26), [0046] 【００７２】である。 [0047] 08-05-2019 11 The error e '(n) in the mode region can be obtained by subtracting the mode amplitude a (n) given by equation (18) from the mode amplitude d' (n) of the target response given by equation (26) . Therefore, operation unit 110 [0048] An error e '(n) in the mode region is derived by the operation given by. In equation (28), [0049] 【００７７】である。 [0050] Next, the mode region error weighting unit 112 weights the errors e ′ (n) (e′0 (n) to e′N−1 (n)) of the mode region to select the mode to be controlled. Weighting is performed using a coefficient B (b0 to bN'-1). The domain conversion filter 114 multiplies the error of the weighted mode domain by the mode eigenfunction Ψ to calculate the error e (n) in the time domain. The weighting for the error e '(n) in the mode domain and the conversion of the weighted mode domain error into the time domain error are [0051] Given by In equation (30), [0052] 【００８３】である。 08-05-2019 12 [0053] Here, when the instantaneous power e (n) T e (n) of the error vector e (n) in the time domain is partially differentiated with the filter coefficient w to obtain an instantaneous estimated value of the gradient vector of the error characteristic surface, [0054] 【００８６】となる。 Therefore, the updating of the coefficients of the control filter 102 is performed by the following equation. [0055] Here, μ is a step size parameter of LMS algorithm (a coefficient for controlling the magnitude of correction at each repetition). [0056] Next, the detailed configuration of the sound field control device of the first embodiment will be described. FIG. 5 is a diagram showing an entire configuration of the sound field control device according to the first embodiment. As shown in the figure, the sound field control apparatus 100 includes a control filter 102 including M adaptive filters with a tap number I, M speakers 104, K microphones 106, mode division filters 108, N ′ , An N 'number of mode region error weighting units 112, an area conversion filter 114, a target response setting unit 116, a mode division filter 118, a filtered x unit 120, and an LMS algorithm processing unit 122. 08-05-2019 13 [0057] The control filter 102, the speaker 104, the microphone 106, the mode division filter 108, the arithmetic unit 110, the mode region error weighting unit 112, and the region conversion filter 114 perform the operation described with reference to FIG. [0058] The target response setting unit 116 is set to have a characteristic (target response characteristic H) corresponding to the sound field space to be reproduced, for example, a characteristic having a delay time about half the number of taps of the filter constituting the control filter 102. The mode division filter 118 derives N ′ number of mode amplitudes from the target response signal output from the target response setting unit 116, and outputs the N ′ mode amplitudes to the calculation unit 110. [0059] The filtered x unit 120 is a filter for creating a reference signal from the input signal u (n). Specifically, the filtered x unit 120 is configured by connecting in series filters having the abovementioned characteristics of Ψ, 、 −1, B, and 上述. The LMS algorithm processing unit 122 performs control according to the equation (34) described above based on the time domain error signal e (n) output from the region conversion filter 114 and the reference signal output from the filtered x unit 120. The filter coefficients of the adaptive filter constituting the filter 102 are adjusted. [0060] As described above, it is possible to correct the transfer characteristic of the entire acoustic space by mode-resolving the sound pressure distribution and controlling the mode having a large amplitude, that is, the mode that adversely affects the transfer characteristic of the acoustic space. [0061] 08-05-2019 14 Next, a modification of the sound field control device of the first embodiment will be described. As shown in FIG. 2, normally, the mode amplitude becomes larger as the value is lower, except for the zero-order. Therefore, by controlling only the low-order mode, it is possible to realize almost the desired acoustic characteristics and to reduce the amount of processing. However, as can be seen from FIG. 2, since the high-order mode signal excluded from the control target contains a large amount of high-frequency components, excluding this high-order mode signal itself reduces the high-frequency components, which is preferable. Absent. For this reason, it is preferable to input the input signal u (n) itself so that at least one of the M speakers 104 also functions as an uncontrolled sound source. [0062] FIG. 6 is a view showing a modification of the sound field control device of the first embodiment, and shows the configuration of the sound field control device which performs control only in the low-order mode. The sound field control device 150 shown in the figure directly outputs the input signal u (n) from the speaker 104 without passing through the control filter 102, and the delay unit 152 is provided to adjust the delay amount at that time. It is equipped. In the delay unit 152, the delay time obtained by subtracting the delay time when passing through the acoustic system C from the delay time set in the target response setting unit 116 is set as the delay amount β. Further, in the mode area error weighting unit 112, for example, only the weighting coefficient bm of the mode to be controlled is set to 1 and the others to 0, and only the error signal of the mode to be controlled is input to the area conversion filter 114. Control by the control filter 102 is performed only for the mode of the unit. [0063] As described above, by performing control only in a part of the modes and outputting the input signal as it is from the speaker 104 in the other modes, it is possible to realize a sound field with less variation in sound pressure due to movement of the listening position. Moreover, the amount of computation can be reduced. [0064] (2-2) Sound Field Control Device of Second Embodiment Having Algorithm for Operating Adaptive Filter in Mode Region The sound field control device of the first embodiment described 08-05-2019 15 above has an algorithm for operating the adaptive filter in the time domain. Although it has, it may be made to operate according to an algorithm which operates an adaptive filter in a mode domain. In order to operate in the mode region, the error calculated in the mode region may be used as it is for updating the coefficients of the adaptive filter. [0065] 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, in the sound field control device of this embodiment, an acoustic system modeling filter 202 that simulates an acoustic system C, and N 'modes from signals (sound pressure) output from the acoustic system modeling filter 202 Mode division filter 204 for deriving amplitude, control filter 206 including N 'adaptive filters with tap number I, and domain conversion for converting a signal of mode domain output from control filter 206 into a time domain signal A filter 208, an acoustic system inverse filter 210 for restoring the acoustic system C ^ simulated by the acoustic system modeling filter 202, M speakers 212, K microphones 214, and sound pressure N 'of the microphone 214 Mode division filters 216 for deriving the mode amplitudes, N ′ operation units 218 for calculating the error of each mode, and And a N 'pieces of mode region error weighting unit 220 for weighting the difference. [0066] When trying to operate the adaptive filter in the mode region, since the coefficients of the control filter 206 are obtained in the mode region, the input signal to the control filter 206 must be a signal in the mode region. For this reason, the input signal u (n) is once passed through the acoustic system modeling filter 202 having characteristics equivalent to the actual acoustic system C, and then the time domain output from the acoustic system modeling filter 202 by the mode division filter 204 Signals are converted to signals in the mode area. [0067] In addition, when sound is actually output from the speaker 212, the signal input to the speaker 08-05-2019 16 212 must be a time domain signal. Therefore, the region conversion filter 208 converts the signal in the mode region output from the control filter 206 into a signal in the time domain again. In addition, since the signal in the time domain output from the domain conversion filter 208 is a signal after passing through the acoustic system C ^ by the acoustic system modeling filter 202 (a signal corresponding to the position of the microphone 214), By passing through the reverse filter 210, the signal corresponding to the position of the speaker 212 is restored. [0068] By the way, the k-th output signal pk (n) of the acoustic system modeling filter 202 modeling the acoustic system C is a convolution of the input signal u (n) and the acoustic system modeling filter 202, [0069] It is represented by Rewriting equation (35) in matrix form, [0070] 【０１０５】となる。 In these equations (35) and (36), [0071] 【０１１０】である。 [0072] The mode amplitude a ^ (n) of the modeling filter output is 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 be able to. 08-05-2019 17 Thus, mode division filter 204 [0073] The mode amplitude a ^ (n) is derived by the operation given by. In equation (42), [0074] 【０１１５】である。 This mode amplitude a ^ (n) is an input signal of the control filter 206. Therefore, the output signal y (n) of the control filter 206 is [0075] 【０１１７】となる。 In equation (44), [0076] 【０１２０】である。 Formula (44) is [0077] It can also be rewritten as follows. In equation (47), [0078] 【０１２６】である。 'n'-1-1 is a vector consisting of elements of the n'th line of the inverse mode eigenfunction' -1. 08-05-2019 18 [0079] Next, the domain conversion filter 208 converts the output signal y (n) of the control filter 206, which is a signal in the mode domain, into a signal in the time domain by applying the mode specific function Ψ. Furthermore, since the signal in this time domain is a signal simulated to the acoustic system C ^ by the acoustic system modeling filter 202, the acoustic system inverse filter 210 applies the inverse filter F of the acoustic system C ^ and restores it. . Therefore, the output signal y '(n) of the acoustic system inverse filter 210 is [0080] 【０１２９】となる。 ここで、 [0081] 【０１３２】である。 [0082] The output signal y ′ (n) is input to the speaker 212, the sound is emitted to the onedimensional sound field of the acoustic system C, and is taken in by the microphone 214. The sound pressure p (n) at the microphone 214 is [0083] Given by ここで、 [0084] 【０１３９】である。 08-05-2019 19 [0085] 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 the equation (7). Thus, mode division filter 216 [0086] The mode amplitude a (n) is derived by the operation represented by ここで、 [0087] 【０１４４】である。 [0088] On the other hand, the mode amplitude d '(n) of the target response is the same as in equation (26), [0089] It is given by ここで、 [0090] 【０１５１】である。 [0091] 08-05-2019 20 The error e '(n) in the mode region can be obtained by subtracting the mode amplitude a (n) given by equation (58) from the mode amplitude d' (n) of the target response given by equation (60) . Therefore, operation unit 218 [0092] An error e '(n) in the mode area is calculated by performing the operation given by. ここで、 [0093] 【０１５６】である。 [0094] Next, the mode region error weighting unit 220 weights the error e ′ (n) of the mode region with the weighting coefficient B according to the following equation (67). [0095] 【０１５９】ここで、 [0096] 【０１６２】である。 [0097] If 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 to obtain an instantaneous estimated value of the gradient vector of the error characteristic surface, 08-05-2019 21 [0098] 【０１６５】となる。 Therefore, updating of the coefficients of the control filter 206 is performed by the following equation. [0099] Here, μ is a step size parameter of the LMS algorithm, which is a coefficient for controlling the magnitude of correction in each repetition. [0100] Next, the detailed configuration of the sound field control device of the second embodiment will be described. FIG. 8 is a diagram showing an entire configuration of a sound field control device according to the second embodiment. As shown in the figure, the sound field control apparatus 200 includes an acoustic system modeling filter 202, a mode division filter 204, a control filter 206 including N adaptive filters with the number of taps I, a region conversion filter 208, and an acoustic system inverse. Filter 210, M speakers 212, K microphones 214, mode division filter 216, N ′ arithmetic units 218, N ′ mode region error weighting units 220, target response setting unit 222, mode division filter 224, A filtered x unit 226 and an LMS algorithm processing unit 228 are provided. [0101] The acoustic system modeling filter 202, the mode division filter 204, the control filter 206, the area conversion filter 208, the acoustic system inverse filter 210, the speaker 212, the microphone 214, the mode division filter 216, the arithmetic unit 218, and the mode area error weighting unit 220 The respective operations described in FIG. 7 are performed. 08-05-2019 22 [0102] The target response setting unit 222 sets a characteristic (target response characteristic H) corresponding to the sound field space to be reproduced, for example, a characteristic having a delay time about half the number of taps of the filter constituting the acoustic system inverse filter 210. There is. The mode division filter 224 derives N ′ number of mode amplitudes from the target response signal output from the target response setting unit 222, and outputs the N ′ mode amplitudes to the calculation unit 218. [0103] The filtered x unit 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 unit 226 is configured by connecting in series filters having the respective characteristics of Ψ, ^, F, Ψ−1, and B described above. Based on the error signal e (n) of the mode region output from the mode region error weighting unit 220 and the reference signal output from the filtered x unit 226, the LMS algorithm processing unit 228 follows the equation (71) described above. The filter coefficients of the adaptive filter constituting the control filter 206 are adjusted. [0104] As described above, by performing control by the control filter 206 in the mode region, it is possible to control the mode with large amplitude, that is, the mode that adversely affects the transfer characteristic of the acoustic space, and correct the transfer characteristic of the entire acoustic space. It becomes possible. [0105] 08-05-2019 23 Next, a modification of the sound field control device of the second embodiment will be described. The sound field control apparatus according to the present embodiment may control only a part of the modes (in particular, the lower order modes), as in the first embodiment in which control is performed in the time domain. [0106] FIG. 9 is a view showing a modified example of the sound field control device of the second embodiment, and shows the configuration of the sound field control device which performs control for only a part of the modes. The sound field control device 250 shown in the figure bypasses from the acoustic system modeling filter 202 to the acoustic system inverse filter 210 and outputs the input signal u (n) directly from the speaker 212, and adjusts the delay amount at that time. In order to do this, a delay 252 is provided. In the delay unit 252, the delay time obtained by subtracting the delay time of the acoustic system C from the delay time set in the target response setting unit 222 is set as the delay amount β. Further, in the control filter 206, for example, the tap coefficient corresponding to the high-order mode not to be controlled is set to 0, and the filter operation and the control using the operation result are performed only for a part of the modes. ing. [0107] FIG. 10 is a view showing a modification of the sound field control device shown in FIG. Like the sound field control device 260 shown in the same figure, with respect to the mode components not to be controlled, control is performed similarly to only a part of the modes by passing an amplifier 262 of coefficient 1 without passing through the control filter 206. The purpose of the can be achieved. [0108] (2-3) Sound Field Control Device of the Third Embodiment Having an Algorithm for Operating an 08-05-2019 24 Adaptive Filter in the Mode Region The sound field control device of the second embodiment described above needs to be provided with the acoustic system inverse filter 210 . For this reason, a preparatory procedure of calculating the acoustic system inverse filter 210 when configuring the system takes one step more. Also, depending on the fluctuation of the acoustic system C, an error may occur in the inverse filter F of C ^, and accurate control may not be possible. Therefore, in the sound field control apparatus of the third embodiment, the area conversion filter 208 and the acoustic system inverse filter 210 used in the sound field control apparatus of the second embodiment are taken into the control filter 206. [0109] 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, in the sound field control device of this embodiment, an acoustic system modeling filter 302 that simulates an acoustic system C, and N 'modes from signals (sound pressure) output from the acoustic system modeling filter 302 Sound pressure of mode division filter 304 for deriving amplitude, control filter 306 including N '× M adaptive filters having the number of taps I, M speakers 312, K microphones 314, and microphones 314 , N ′ operation units 318 for calculating the error of each mode, and N ′ mode region error weighting units 320 for weighting the errors of each mode. And have. [0110] The control filter 306 included in the sound field control device of the present embodiment is obtained by incorporating the area conversion filter 208 and the acoustic system inverse filter 210 in the control filter 206 shown in FIG. Therefore, first, the output signal y '(n) of the acoustic system inverse filter 210 shown in FIG. [0111] 【０１８０】となる。 Here, assuming that FΨW is rewritten to W again, that is, the control filter 206 shown in FIG. 7 etc. and the area conversion filter 208 and the acoustic system inverse filter 210 are taken as the control filter 306, the control filter 306 The output signal y '(n) of [0112] 08-05-2019 25 Given by In equation (73), [0113] 【０１９３】である。 [0114] The output signal y ′ (n) is input to the speaker 312, the sound is emitted to the onedimensional sound field of the acoustic system C, and is captured by the microphone 314. The sound pressure p (n) at the microphone 314 is [0115] It is given by ここで、 [0116] 【０２００】である。 [0117] The mode amplitude a (n) can be obtained by performing mode decomposition on the sound pressure p (n) at the microphone 314 obtained by the equation (84) in the same manner as the equation (7). Thus, mode division filter 316 [0118] 08-05-2019 26 The mode amplitude a (n) is derived by the operation given by ここで、 [0119] 【０２０５】である。 [0120] On the other hand, the mode amplitude d '(n) of the target response is the same as in equation (26), [0121] Given by ここで、 [0122] 【０２１３】である。 [0123] The error e '(n) in the mode region can be obtained by subtracting the mode amplitude a (n) given by equation (88) from the mode amplitude d' (n) of the target response given by equation (90) . Therefore, operation unit 318 [0124] 08-05-2019 27 An error e '(n) in the mode area is calculated by performing the operation given by. ここで、 [0125] 【０２１８】である。 [0126] Next, the mode region error weighting unit 320 weights the error e ′ (n) of the mode region with the weighting coefficient B. This weighting is [0127] Calculated by: In equation (97), [0128] 【０２２４】である。 [0129] Here, 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, the instantaneous estimated value of the gradient vector of the error characteristic surface is determined [0130] 【０２２７】となる。 08-05-2019 28 Therefore, the updating of the coefficients of the control filter 306 is performed by the following equation. [0131] Here, μ is a step size parameter of the LMS algorithm, which is a coefficient for controlling the magnitude of correction at each repetition. [0132] Next, the detailed configuration of the sound field control device of the third embodiment will be described. FIG. 12 is a diagram showing a detailed configuration of the sound field control device of the third embodiment. As shown in the figure, the sound field control apparatus 300 includes an acoustic system modeling filter 302, a mode division filter 304, a control filter 306 including N '× M adaptive filters with a tap number I, M speakers 312, K microphones 314, mode division filters 316, N 'arithmetic units 318, N' mode region error weighting units 320, target response setting units 322, mode division filters 324, filtered x units 326, LMS algorithm processing A section 328 is provided. [0133] The acoustic system modeling filter 302, the mode division filter 304, the control filter 306, the speaker 312, the microphone 314, the mode division filter 316, the arithmetic unit 318, and the mode region error weighting unit 320 perform the operations described with reference to FIG. [0134] The target response setting unit 322 is set to a characteristic (target response characteristic H) corresponding to the sound field space to be reproduced, for example, a characteristic having a delay time about half the number of taps of the filter constituting the control filter 306 . 08-05-2019 29 The mode division filter 324 derives N ′ number of 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. [0135] The filtered x unit 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 unit 326 is configured by connecting in series filters having characteristics of C, Ψ−1, and B. The LMS algorithm processing unit 328 generates the error signal e (n) of the mode region output from the mode region error weighting unit 320 and the reference signal output from the filtered x unit 326 according to the equation (101) described above. The filter coefficients of the adaptive filter constituting the control filter 306 are adjusted. [0136] As described above, by using the control filter 306 incorporating the area conversion filter 208 and the acoustic system inverse filter 210, it is not necessary to obtain the inverse filter F in advance, so that the preparation procedure of the system can be reduced by one step. Further, since the inverse filter F is incorporated in the control filter, it is possible to cope with the fluctuation of the acoustic system C to some extent. [0137] Next, a modification of the sound field control device of the third embodiment will be described. The sound field control device according to the present embodiment may control only a part of the modes (in particular, the low-order mode) as in the first and second embodiments described above. [0138] 08-05-2019 30 FIG. 13 is a view showing a modification of the sound field control device according to the third embodiment, and shows a configuration of the sound field control device which performs control for only a part of modes. The sound field control device 350 shown in the figure bypasses from the acoustic system modeling filter 302 to the control filter 306 and directly outputs the input signal u (n) from the speaker 312, and adjusts the delay amount at that time. A delay device 352 is provided for this purpose. In the delay unit 352, a delay time obtained by subtracting the delay time of the acoustic system C from the delay time set in the target response setting unit 322 is set as the delay amount β. Also, in the control filter 306, for example, the tap coefficient corresponding to the high-order mode not to be controlled is set to 0, and the filter operation and the control using the operation result are performed only for a part of the modes. It has become. [0139] (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 the optimum arrangement method thereof will be described. Formula (2) mentioned above is mode amplitude of each mode at the time of mode-resolving sound pressure distribution. In the equation (2), there are two controllable variables: qm (ω) which is an input signal to the m-th speaker and lm which is the position of the m-th speaker. Therefore, for example, when canceling the n'th mode, that is, when the mode amplitude of the n'th mode is set to 0, [0140] It must be established. [0141] However, if the speaker is arranged at a position where the term of cos (n'πlm / L) is 0 in the equation (102), the value of the input signal qk (ω) to the speaker may be determined by this speaker Since mode control becomes impossible, the speaker can not be arranged at such a position. Also, if the loudspeaker is placed at a position where cos (n'πlm / L) is as close to 1 as possible, the input signal qk (ω) has a large effect on the control of the mode amplitude, ie, the input signal qk for control. Since the efficiency of (ω) is high, it is desirable to arrange the speaker at such a position. 08-05-2019 31 [0142] In addition, 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 (same for other modes) is There is a peak dip. Therefore, as described above, in order to achieve a flat sound pressure distribution in the entire acoustic space, it is necessary to leave the zero-order mode and cancel the other modes. [0143] Assuming that n '= 0 in the equation (2) to obtain the mode amplitude of the zero-order mode, [0144] Is obtained. Since this equation (103) does not include the term cos (n'πlm / L), the mode amplitude of the zero-order mode is the sum of the input signals qk (ω) to the respective speakers regardless of the positions of the speakers. Indicates that it depends only on. That is, when the input signals qk (ω) to the respective speakers have positive and negative discrete values, they cancel each other when they are summed, and the amplitude of the zero-order mode becomes small. Therefore, the input signals qk (ω) to the respective speakers must always have the same sign. In order to cancel the other modes under this condition, it is necessary to place the loudspeakers at all positions where cos (n'.pi.lm / L) of the other modes has an opposite sign. [0145] 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 embodiment described above, 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 considered in the same manner. The wave equation in the case of a three-dimensional sound field uses the following equation (104) instead of the equation (1) described above. 08-05-2019 32 [0146] Where x 1, x 2 and x 3 are the vertical, horizontal and height positions of the microphone, ω is the angular frequency, p (x 1, x 2, x 3, ω) is the sound pressure and q m is the m th speaker Input signals, l1m, l2m and l3m are the positions of the m-th speaker in the vertical, horizontal and height directions, M is the total number of speakers, ξn'1, ξn'2, , N'2 and n'3 modes of attenuation ratio at the wall, N 'is the total number of modes, L1, L2 and L3 are the length, width and height of the sound field, ωn'1, n '2, n' 3 (= π c 0 {(n '1 / L 1) 2 + (n' 2 / L 2) 2 + (n '3 / L 3) 2}) are the specific frequencies of the sound field, ρ 0 is the air The density is indicated by c0, the speed of sound. [0147] Also, in equation (104), [0148] 【０２５１】である。 [0149] 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 largely changes when the listening position moves is reduced. Since the noise can be canceled or canceled, the transfer characteristic can be corrected over the entire acoustic space by a small number of speakers and an adaptive filter without particularly increasing the control points, and a flat sound pressure distribution can be realized. [0150] In this case, by arranging the plurality of speakers at a position other than the position corresponding to the vibration node of the mode to be controlled, various control such as reducing or canceling the mode amplitude of the mode is possible. Become. Moreover, it is preferable to arrange each of the plurality of speakers in all positions where the signs of the vibration of the mode to be canceled are opposite, while aligning and outputting the sign of the input signal. Leaving only the other desired modes. 08-05-2019 33 [0151] Brief description of the drawings [0152] 1 is a diagram showing a specific example of a mode decomposition unit configured by applying the mode decomposition method. [0153] 2 is a diagram showing the frequency characteristics of each mode included in the acoustic system. [0154] 3 is a diagram showing an amplitude state of the mode. [0155] 4 is a diagram showing a schematic configuration of the sound field control device of the first embodiment. [0156] 5 is a diagram showing the overall configuration of the sound field control device of the first embodiment. [0157] 6 is a diagram showing a modification of the sound field control device of the first embodiment. [0158] 7 is a diagram showing a schematic configuration of a sound field control device of the second embodiment. [0159] <Figure 8> It is the figure which shows the entire constitution of the sound field control control equipment of 2nd execution form. 08-05-2019 34 [0160] <Figure 9> It is the figure which shows the deformation example of the sound field control control equipment of 2nd execution form. [0161] 10 is a diagram showing a modification of the sound field control device shown in FIG. [0162] <Figure 11> It is the figure which shows the outline constitution of the sound field control control equipment of 3rd execution form. [0163] <Figure 12> It is the figure which shows the detailed constitution of the sound field control control equipment of 3rd execution form. [0164] <Drawing 13> It is the figure which shows the deformation example of the sound field control control equipment of 3rd execution form. [0165] <Figure 14> It is the figure which shows the constitution of the adaptive equalization system which is applied to the audio device. [0166] Explanation of sign [0167] Reference Signs List 100 sound field control unit 102 control filter 104 speaker 106 microphone 108, 118 mode division filter 110 arithmetic unit 112 mode area error weighting unit 114 area conversion filter 116 target response setting unit 120 filtered x unit 122 LMS algorithm processing unit 08-05-2019 35 08-05-2019 36

1/--страниц