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JP2000295697

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DESCRIPTION JP2000295697
[0001]
BACKGROUND OF THE INVENTION 1. Field of the Invention The present invention relates to a
directional loudspeaker and, more particularly, to a directional loudspeaker capable of listening
to sounds of different signal sources in different areas with respect to a pair of speakers.
[0002]
2. Description of the Related Art Conventionally, as a device for enabling different sounds to be
heard simultaneously in different areas of the same space, the directivity of sound radiated from
a speaker is controlled in two or more desired directions. There was something. For example, as
shown in FIG. 1 (1), each input signal (AS) supplied to an array speaker (AS) in which a plurality
of speakers are linearly arranged is disclosed in Japanese Unexamined Patent Publication No. 6205496 (“Speaker device”). The digital signals (IN1, IN2) are individually processed by digital
filters (DF1 to m) so as to make the delay time and the like different, and thereby directivity (D1)
of each input signal sound emitted from the array speaker (AS) , D2) have been described, but
this technique has the disadvantage of a large scale of the device.
[0003]
In addition, as shown in FIG. 1 (2), in JP-A-5-344579 ("Speaker system and television receiver
using the speaker system"), the speaker vibration of two speaker units (SU) is shown. A
technology is described in which an acoustic partition (PP) and a sound absorbing panel (AP) are
provided on an acoustic pipe (ST) for guiding a sound wave emitted from a plate to give
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directivity of a speaker in any direction in structure. However, this technique has the
disadvantage that the suppression ratio effect is small and the switching of the pointing direction
is not easy.
[0004]
Further, in Japanese Patent Laid-Open No. 5-333878 ("Crosstalk sound control device"), when a
plurality of pairs of speaker systems consisting of two stereo speakers are juxtaposed to listen to
different sounds for each pair, A technique has been described that reduces crosstalk noise and
allows only the desired sound to be heard in each pair of areas by synthesizing cancellation
sound between adjacent speaker systems by digital signal processing. However, this technique
has the disadvantage that the canceling area (volume) is small.
[0005]
On the other hand, the inventors of the present invention have two people (or two groups) in
different areas of the same space by canceling predetermined sound wave components emitted
from two closely arranged speakers by digital signal processing. Japanese Patent Application No.
10-39378 has proposed a speaker device in which the listener of the speaker can simultaneously
hear the sound of different sound sources (this proposal is hereinafter referred to as "the
proposal".
However, in this speaker device, there is a disadvantage that the performance may be influenced
by the installation place or surrounding conditions.
[0006]
SUMMARY OF THE INVENTION In view of the above problems, the present invention makes it
difficult to listen to the radiation emitted from the speakers in a certain area in a space where the
radiation emitted from the pair of speakers is heard. In order to be able to simultaneously listen
to the sound of different sound sources in different areas, and to be able to arbitrarily and
optimally change such hard-to-listen areas and individually listenable areas, An object of the
present invention is to provide a directional loudspeaker system with further improved
directivity.
[0007]
SUMMARY OF THE INVENTION According to one aspect of the present invention, a sound wave
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is emitted which is juxtaposed at an interval of 1/8 wavelength to 1 wave of sound wave in the
frequency domain of an acoustic signal and contains the acoustic signal component. It is
respectively interposed between two speakers (6a, 6b), an acoustic signal source (1b or 1a) and
one speaker (6a or 6b), and is emitted from both speakers in the listening space for both
speakers A filter (3a or 3b) for canceling out sound waves by interference between sound waves,
a sensor microphone (7) arranged in front of both speakers, and a filter for adjusting the filter
coefficient of the filter based on the detection output of this sensor microphone A directional
loudspeaker system is provided which comprises adjusting means (9a or 9b).
[0008]
According to another feature of the invention, the sound waves are emitted side by side at an
interval of 1⁄8 wavelength to 1 wave length of the sound wave in the frequency domain of the
first and second sound signals and including the first and second sound signal components. The
first and second speakers (1b, 1a) and the second and first acoustic signal sources (1b, 1a) and
the first and second speakers are respectively interposed and different in the listening space for
both speakers In the region, first and second filters (3a, 3b) for canceling the first and second
acoustic signal components in the sound waves radiated from both speakers by the interference
between the sound waves, respectively, and disposed in front of both speakers There is provided
a directional loudspeaker system comprising a sensor microphone (7) and filter adjustment
means (9a, 9b) for adjusting the filter coefficients of the first and second filters based on the
detection output of the sensor microphone. .
[0009]
[Operation] According to one aspect of the present invention, two loudspeakers are closely
arranged through an interval of 1/8 wavelength of sound wave of controlled frequency domain
to 1 wavelength, and the sound wave of the acoustic signal in this frequency domain is arranged.
Emit
A digital filter is interposed between the acoustic signal source and one of the speakers so that
the sound waves emitted from the two speakers are canceled out by the interference of the
sound waves in a certain region of the listening space of the two speakers.
Then, the filter coefficient of this digital filter is controlled to an optimum value corresponding to
the sensor microphone direction or position by the filter adjuster that operates in accordance
with the detection output from the sensor microphone, thereby achieving cancellation of the
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sound wave. Can be arbitrarily controlled according to the position of the sensor microphone.
[0010]
According to another feature of the invention, two loudspeakers are arranged in close proximity
via a spacing of 1/8 wavelength to 1 wavelength of the sound wave of the controlled frequency
domain, mainly in this frequency domain, each of the content types Sound waves including
different, ie uncorrelated first and second acoustic signal components are emitted from both
speakers.
First and second digital filters are respectively interposed between the second and first acoustic
signal sources and the first and second speakers, and each of the speakers is different in a
different area within the listening space of both speakers. The second or first acoustic signal
component of the sound wave emitted from the light source is canceled by the interference
between the sound waves.
Then, the filter coefficient of this digital filter is controlled to an optimum value corresponding to
the sensor microphone direction or position by the filter adjuster that operates in accordance
with the detection output from the sensor microphone, thereby achieving cancellation of the
sound wave. The direction or the position can be arbitrarily controlled according to the position
of the sensor microphone for each area.
[0011]
In the present invention, by thus using the error signal from the sensor microphone in the
vicinity of the speaker, the position at which cancellation can be achieved, in particular, the
direction to the speaker can be arbitrarily controlled, so the device can be miniaturized.
Nevertheless, any directivity effect can be obtained, and the direction and position of the
muffling can be easily switched. Therefore, in the listening space, it is possible to make it difficult
to hear the sound emitted from the speaker in a certain area, or to simultaneously and separately
hear the sound of different sound sources in different areas. Furthermore, it is possible to
arbitrarily change such hard-to-listen areas and individually-listenable areas. Moreover,
according to the experiment, it was confirmed that the suppression ratio can be set to 15 dB or
more, and a good directivity effect can be obtained in a wide area of 0.5 to 1 m2.
03-05-2019
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[0012]
BEST MODE FOR CARRYING OUT THE INVENTION Preferred embodiments of the present
invention will now be described in detail with reference to the drawings. The following
embodiments are merely examples, and various modifications can be made without departing
from the spirit of the present invention.
[0013]
System Configuration Referring to FIG. 2, there is shown a block diagram illustrating the
configuration of a directional loudspeaker system according to an embodiment of the present
invention. In this example, the directional loudspeaker apparatus includes sound sources 1a and
1b that generate two different first and second sound signals Sa and Sb, and the sound signals Sa
and Sb output from the sound sources 1a and 1b are respectively It is supplied to the delay units
2a and 2b and to the first and second digital filters 3b and 3a. Here, the delay units 2a and 2b
are configured to delay the acoustic signals Sa and Sb by a time approximately equal to the signal
processing time in the digital filters 3a and 3b, respectively.
[0014]
The output signals of the delay 2a and the first digital filter 3a are input to the adder 4a, and the
output signals of the delay 2b and the second digital filter 3b are input to the adder 4b. The
addition output signals of the adders 4a and 4b are amplified by the amplifiers 5a and 5b,
respectively, and then input to the first and second speakers 6a and 6b of the speaker device 6,
respectively. The speaker device 6 has the two speakers 6a and 6b arranged at an interval (d0) of
1⁄8 wavelength to 1 wavelength of the sound wave of the main frequency range of the first and
second acoustic signals Sa and Sb. The signals input from the amplifiers 5a and 5b are emitted as
sound waves from the respective speakers.
[0015]
In one embodiment of the present invention, the first and second digital filters 3a and 3b are
respectively the first and second sound waves radiated from the two speakers 6a and 6b in
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different regions in the space for the two speakers 6a and 6b. It has a function of canceling the
second acoustic signal component by the interference of sound waves, and an error microphone
7 is attached near the main body of the speaker device 6 by the beam 6BM. The error
microphone 7 outputs an error signal according to the acoustic state of the installation
environment when the system of FIG. 2 is installed.
[0016]
This error signal is input to the first and second filter adjusters 9a and 9b via the changeover
switch element 8, and the filter adjusters 9a and 9b set the filter coefficients of the digital filters
3a and 3b according to the error signal. Adjust to the optimum value and perform adaptive filter
control. As a result, an optimal digital filter coefficient can be obtained according to the acoustic
state at the time of installation of the device, and the region to be canceled can be changed
according to the installation direction of the error microphone 7. This will be specifically
described below.
[0017]
It should be noted that monaural sound can be introduced instead of the first and second
acoustic signals Sa and Sb from the respective signal sources 1a and 1b, and the filter coefficient
is always optimized using the error microphone 7 It is also possible to cause the operation to be
rewritten. By using it in this way, it is possible to always adapt to changes in room temperature
and changes in the acoustic transfer function with respect to the movement of a person or fixture
in the room.
[0018]
Now, consider a point Pa as shown in FIG. 2 in the sound listening space for both speakers 6a
and 6b, and the sound to be heard at this point Pa is the sound emitted from the first and second
speakers 6a and 6b. The sound pressure due to is linearly superimposed. Therefore, the sound
pressure A (ω) at the listening point Pa is X (ω) of the voltage of the first acoustic signal Sa
output from the sound source 1a and Y of the second acoustic signal Sb output from the sound
source 1b. Let (ω) be (ω is the angular frequency of the sound signal) and can be expressed by
the following equation (1): A = (αX + βY) + (γX + δY) (1) where α is the circuit element 1a →
Transfer function between elements 1a to 6a to Pa with 2a → 4a → 5a → 6a and space 6a → Pa
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as path, β is path from circuit element 1b → 3a → 4a → 5a → 6a and space 6a → Pa A transfer
function between elements 1b to 6a to Pa, γ is a circuit element 1a → 3b → 4b → 5b → 6b, and
a space 6b → Pa is a transfer function δ between 1a to 6b to Pa is a circuit element 1b →
Between 1b to 6b to Pa, with 2a → 4a → 5b → 6b and space 6b → Pa as a route It is a transfer
function.
[0019]
That is, the first term on the right side represents the sound emitted from the first speaker 6a
and heard at one point Pa in the area 7a, and the second term on the right side is the sound
emitted from the second speaker and heard at the same point Pa Represents When the digital
coefficient of the second digital filter 3b is adjusted to change the coefficient β and β = −δ,
equation (1) is obtained as equation (2) and the term including the second acoustic signal voltage
Y is eliminated can do. A = αX + γX = (α + γ) X (2)
[0020]
At the listening point Pa, this meaning is referred to as a sound (hereinafter referred to as "first
acoustic signal component") derived from the first acoustic signal Sa (signal voltage X) from the
sound source 1a. Only), and the crosstalk sound [(β + δ) Y] derived from the second acoustic
signal Sb (signal voltage Y) from the sound source 1b can not be heard.
[0021]
Further, the same operation is performed to adjust α = −γ also to the filter of the first digital
filter 3 a with respect to another one point Pb which is symmetrical with respect to the center
line between the two speakers 6 a and 6 b at the point Pa. Thus, at the listening point Pb, a sound
(hereinafter referred to as “second sound signal component”) derived from the second sound
signal Sb (signal voltage Y) from the sound source 1 b. Only) and can not hear the crosstalk
sound [(.alpha. +. Gamma.) X] derived from the first acoustic signal Sa (signal voltage X) from the
sound source 1a. The operation up to this point can be said to be similar in principle to the one
according to the above-mentioned JP-A-5-333878.
[0022]
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7
In such a configuration, it is only one point in the listening space that can satisfy .beta. =-. Delta.
Or .alpha. =-. Gamma. By the digital filters 3b and 3a, and this point is called "cancellation point".
)においてクロストークはゼロとなる。 In addition, around the cancellation point, an area with
less crosstalk is generated.
[0023]
For example, when the crosstalk at the listening point Pa is made zero by adjusting the filter
coefficient of the second digital filter 3b to establish β = −δ, the crosstalk is reduced in the
hatched area A in FIG. When the crosstalk at the listening point Pb is made zero by adjusting the
filter coefficient of the first digital filter 3a, the crosstalk is reduced in the region B opposite to
this. This is clarified by the present inventors, as will be described below, and the invention
utilizing this phenomenon is described in detail in the above proposal.
[0024]
Generally, when the speakers are installed sufficiently apart in the diffuse sound field, the space
where the crosstalk is -10 dB is almost circular and is about 10 times as large as the wavelength
of the sound wave in the control band. (See, for example, “A. David and S. J. Elliot 1993 applied
acoustics 41, 63-79. See “Numerical studies of actively generated quiete zones”). Also,
conversely, when the sound source speakers are brought close to 1/8 or less of the wavelength
of the sound wave of the control frequency, the area where crosstalk is less than -10 dB spreads
in front of both of the two speakers, It has been found that, in the case of, the areas of crosstalk
reduction overlap each other.
[0025]
FIG. 3 shows a hatching in a system where speakers are arranged in parallel as shown in FIG. It is
a figure, and a crosstalk reduction area | region changes like FIG. 3 (1)-FIG. 3 (3) as a function of
a speaker space | interval with respect to predetermined | prescribed wavelength (lambda).
When the speakers are sufficiently separated and the relationship of "speaker distance d1>
wavelength λ" is established, crosstalk is reduced in a substantially circular small area as shown
in FIG. 3 (1). Further, in the case where both speakers are brought close to each other and the
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relationship of "speaker distance d2> (1/8) × wavelength λ", the crosstalk reduction region is
greatly expanded, but the first and second acoustic signal components are Cross talk reduction
areas overlap each other.
[0026]
However, in the case where the speaker spacing d0 is set to (1/8) × λ <d0 <λ (3) as in the
present invention, the crosstalk reduction region of each sound signal component is independent,
and is large. It will be of volume. This has already been described in the above proposal.
[0027]
When the present inventors repeated studies, under the condition shown in FIG. 3 (2), as shown
in FIG. 4, the error microphone 7 is placed at the place P near the speaker and sounds other than
the target sound are By changing the filter coefficient of the first or second digital filter 3a, 3b so
as to cancel, any angle that the point P makes with the center line CL passing through the middle
point O of the first and second speakers 6a, 6b It was found that the directivity O → Pa can be
changed in the crosstalk reduction region according to θ. That is, the crosstalk reduction area
from both the speakers 6a and 6b and the direction thereof change according to the position of
the error microphone 7 as shown by a solid line, a broken line and a dashed line in FIG.
[0028]
For example, as shown in FIG. 4, the error microphone 7 is placed at both the speaker nearby
positions P in the direction of the solid line arrow forming an angle θ from the middle point O,
and the error signal cancels the sound of the second acoustic signal component Sb Adjustment of
the filter coefficient of the first digital filter 3a, the −10 dB line indicating the boundary of the
crosstalk reduction area is represented by La (solid line), and the cancellation point Pa at this
time is approximately the middle point It is in the direction from O to the microphone position P.
When adjusting the filter coefficient value by placing the microphone position P near the speaker
in the direction of the broken line, a good crosstalk reduction area becomes an area surrounded
by the line La ′ (dotted line), and the speaker is placed in the dashed dotted line direction The
crosstalk reduction region is given by a line La ′ ′ (one-dot chain line).
03-05-2019
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[0029]
In addition, the error microphone 7 is placed near each speaker in the opposite symmetrical
direction with respect to the center line CL, and the filter of the second digital filter 3b is used to
cancel (minimize) the sound of the first acoustic signal component Sa. When the coefficients are
adjusted, crosstalk reduction regions represented by lines Lb (solid line), Lb '(dotted line), and Lb
"(one-dot chain line) can be similarly obtained. According to experiments, it was confirmed that a
crosstalk reduction area in a desired direction can be obtained if the microphone position P is 30
cm or more from the center O.
[0030]
Thus, even if the position of the error microphone 7 is not set to the cancellation points Pa and
Pb as described above, it is set to a position closer to the speaker so that an error other than the
target sound is canceled at that position. By feedback from the microphone 7 to the first or
second digital filter 3a, 3b, the same cancellation area setting effect as when each filter 3a, 3b is
set to perform cancellation at the cancellation points Pa, Pb is obtained. It is something that can
be obtained. That is, according to the present invention, the error microphone 7 is used to
optimally adjust the filter coefficients of the digital filters 3a and 3b to generate a crosstalk
reduction region in the microphone position direction, thereby enabling the error microphone 7
to be located near any speaker. The optimum crosstalk reduction area (direction or position) can
be set / adjusted / updated arbitrarily by simply setting.
[0031]
When adjusting this filter coefficient, install the system shown in FIG. 2 inside the room serving
as a listening space or in a laboratory simulating this, and output the test monaural (identical)
acoustic signal from both signal sources 1a and 1b. , Take out the error signal from the error
microphone 7. For example, when the error microphone 7 is provided on the speaker side from
the cancellation point Pa and the switch element 8 is inserted to the a side, the first filter adjuster
9a responds to the error signal from the error microphone 7 as the first digital filter 3a. The filter
coefficients of are adjusted to an optimal value that minimizes the error signal value. Thereby, it
is possible to set or adjust a region in the direction according to the installation position of the
error microphone 7 (for example, the region A in FIG. 2) as the crosstalk reduction region of the
second acoustic signal component. As the adjustment method, it is optimal to output an acoustic
signal from one of the signal sources 1a and 1b (for example, 1b), and to use a digital filter (for
03-05-2019
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example, 3a) for passing the acoustic signal according to the error signal from the error
microphone 7. There are various methods such as adjustment.
[0032]
Also, when the direction of the beam 6BM is switched, for example, positioned on the speaker
side of the cancellation point Pb and the switch element 8 is inserted to the b side, the filter of
the second digital filter 3b is processed by the second filter adjuster 9b. The coefficient is
optimally adjusted in the same manner as described above, and a region in the direction
according to the installation position of the error microphone 7 in this case (for example, region
B in FIG. 2) is set as the crosstalk reduction region of the first acoustic signal component. Can be
set or adjusted.
[0033]
Therefore, for example, when the error microphone 7 is positioned in the direction
corresponding to each of the listening areas La1 to Lam and Lb1 to Lbn shown in FIG. 4 to adjust
the filter coefficient, it corresponds to each of the areas La1 to Lam and Lb1 to Lbn. In this area,
the volume from one signal source can be made lower than that in the surrounding area, or the
respective sounds from different signal sources can be heard in different areas, and such an
inaudible area or The individual sound dedicated listening area can be arbitrarily changed.
[0034]
That is, the gist of the present invention is that it is possible to set, adjust, and update the
coefficients of the digital filter using the signal of the error microphone 7 attached by the beam 9
in the vicinity of the main body of the speaker device 6.
Thereby, not only at the time of installation of the loud-speaking device but also at the time of
use, it is possible to obtain an optimal digital filter coefficient according to the acoustic state of
the listening space.
For example, it is also possible to use the error microphone 7 to always rewrite the filter
coefficient so as to be optimal. By using in this manner, it is possible to always adapt to changes
in room temperature and changes in the acoustic transfer function with respect to movement of a
person or fixture in the room.
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[0035]
In the loudspeaker apparatus according to the present invention, in order to reduce crosstalk in a
wide range as shown in FIG. 3 (2), it is only necessary to use the error signal at the error
microphone 7 installed near the speaker device 6. In other words, the error in the wide spatial
range can be represented by the microphone 7 in the vicinity of the speaker device 6. Further,
the direction of this crosstalk reduction can be varied by switching the angle of the beam 6BM.
[0036]
The above-mentioned digital filter for setting a given transfer function by adaptive filter control
and setting a listening area in an arbitrary direction is a FIR (finite impulse response) filter or IIR
(infinite impulse response) filter well known to those skilled in the art. And so on.
[0037]
FIG. 5 is a view showing another example of the speaker device 6 of the system shown in FIG. 2;
each of the speakers 6a and 6b of FIG. 2 is used as the bass speakers 6a1 and 6b1 and the hightone speakers 6a2 and 6b2, respectively. A system in which outputs of the respective amplifiers
are distributed to the bass speakers 6a1 and 6b1 and the high-pitch speakers 6a2 and 6b2
according to the frequency by the crossover networks 10a and 10b which are separated and
provided between the respective amplifiers 5a and 5b. It is.
[0038]
The speakers 6a1, 6a2, 6b2 and 6b1 are linearly arranged in this order, and for example, the
distance between the centers of the bass speakers 6a1 and 6b1 disposed on the outside is 40 cm,
and the high-sound speakers 6a2 disposed on the inside The distance between the centers of 6b2
is 5 cm.
With such a speaker arrangement, good directivity is obtained in the band of 850 to 6800 Hz for
the inner high-pitched speakers 6a2 and 6b2, and in the band of 106 to 850 Hz for the two outer
low-pitched speakers 6a1 and 6b1. Be
Thus, the loudspeaker configuration of the high bass separation of FIG. 5 provides crosstalk
03-05-2019
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reduction in the desired area with a wide band of 110 to 6800 Hz. Note that the configuration of
the speakers is not limited to the one in which one pair of speakers is provided for each of the
low tone and high tone as shown in FIG. 5, but only one pair is provided by changing the
configuration of the crossover networks 10a and 10b. It is also possible to use speakers or three
or more sets of speakers.
[0039]
The directional loudspeaker apparatus according to the present invention can consciously create,
in any direction or position, a region where the volume of a television broadcast or radio
broadcast is smaller than that in the same room, for example. It can be used as a speaker of a
television audio multiplex broadcast receiver, and any area where two people (or two groups) can
simultaneously enjoy different voices of the main and auxiliary channels of a television audio
multiplex broadcast can be used. Can be consciously created in the direction or position of In
addition, it is not limited to broadcasting such as television and radio, and is used for various
directional loud-sounding systems in the case of creating a small volume in a certain area or
creating different areas in two areas indoors etc. be able to.
[0040]
As described above, according to one aspect of the present invention, two speakers are closely
arranged via an interval of 1/8 wavelength of sound wave of controlled frequency domain to 1
wavelength, A sound wave of an acoustic signal in this frequency range is emitted, and a digital
filter interposed between the acoustic signal source and one of the speakers causes sound waves
emitted from both speakers in a certain area of the listening space of both speakers. Is canceled
by the interference between the sound waves, and the region (position, direction) where this
sound wave cancellation is achieved is a filter that adjusts the filter coefficient of the filter based
on the detection output of the sensor microphone placed in front of both speakers It can control
arbitrarily by changing with an adjustment means.
[0041]
Also according to another feature of the invention, two loudspeakers are arranged in close
proximity via a spacing of 1/8 wavelength of the sound wave of the controlled frequency domain
to 1 wavelength, mainly in this frequency domain, A sound wave having different content types,
that is, uncorrelated first and second sound signal components is emitted from both speakers,
and is interposed between the first and second sound signal sources and the second and first
speakers By the first and second digital filters, the first and second acoustic signal components of
the sound waves radiated from the respective speakers are canceled by the interference of the
03-05-2019
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sound waves in different regions within the listening space of both speakers, The area (position,
direction) where cancellation is achieved is adjusted by the filter adjustment means with the filter
coefficient of each filter based on the detection output of the sensor microphone disposed in
front of both speakers By further, it can be controlled arbitrarily for each of the regions.
[0042]
Therefore, by optimally adjusting the digital filter coefficient based on the detection output of the
sensor microphone disposed in the vicinity of the speaker, the cancellation achievement region,
in particular, the direction with respect to the speaker can be arbitrarily controlled according to
the sensor microphone arrangement direction. In spite of the downsizing of the device, it is
possible to obtain an arbitrary high-performance directivity effect and to easily switch the
direction and position of the muffling.
This makes it difficult to listen to the sound emitted from the speaker in a certain area in the
listening space, or allows different sound sources to be simultaneously heard in different areas,
and further, It is possible to arbitrarily change the hard-to-listen area and the individuallylistenable area.
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