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JPH11168792

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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,
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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
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[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]
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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
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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-
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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,
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[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.
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[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
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[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
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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]
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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。
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[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.
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[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]
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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
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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
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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
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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。
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[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]
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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.
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[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]
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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
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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]
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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]
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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]
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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
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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 .
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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]
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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
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[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
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