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JP2007274131

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DESCRIPTION JP2007274131
An object of the present invention is to perform high-brightness loudness without complex
configuration or signal processing. A plurality of horn microphones 12 and 15 are dispersedly
arranged on a ceiling surface 11. The horn microphones 12 and 15 are respectively constituted
by the microphones 13 and 16 attached to the throats of the horns 13 and 16 and the horns 13
and 16, and the sounds in the cover areas (1) and (2) are collected. And, the sound outside the
cover area has a limited directivity characteristic to reduce. The room is covered comfortably
with such a plurality of horn microphones. The input from each horn microphone is monitored,
the horn microphone with the maximum input level is detected as the sound source position, and
the input from the horn microphone is amplified and amplified from the speaker. By arranging a
plurality of microphones at predetermined intervals in the opening of the horn microphone and
using a simple microphone array, it is possible to obtain uniform directivity characteristics from
low to high frequencies. [Selected figure] Figure 2
Loudspeaker system and sound collector
[0001]
The present invention relates to a hands-free loudspeaker system and a sound collector suitable
for use in the loudspeaker system.
[0002]
If the speaker and the audience are in the same room, and the size of the venue is more than a
certain size, and the voice content of the speaker can not be sufficiently heard only by the voice,
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it is necessary to raise the voice.
In the case of loud-speaking, in order to pick up clear sound, usually a fixed microphone is
installed and it is necessary for the speaker to speak at that position or for the speaker to carry
the microphone Met. Then, when the speaker changes, for example, at the time of question and
answer, it is necessary for the requester to move to the position of the fixed microphone or to
turn the microphone to the speaker. When the speaker has a microphone, if the speaker moves,
the positional relationship between the microphone and the speaker may change, which may
cause howling or coloration. Therefore, in some cases, a specialized operator is required so that
howling does not occur. When the microphone is fixedly installed on the desk, the arrangement
of the microphone changes at the same time as the arrangement of the desk and the chair when
the form of the entertainment in the room is changed, and the setting needs to be performed
again.
[0003]
Therefore, there has been proposed a hands-free loudspeaker system in which a speaker's voice
is collected using a microphone array, amplified, and output from a speaker (Patent Document 1
and Non-Patent Document 1). In a hands-free loudspeaker system (off-microphone loudspeaker
system), the looping of the loudspeaker into the microphone forms a loop whose acoustic
feedback component determines the system gain. Therefore, in the hands-free loudspeaker
system, (1) How can only the direct sound be picked up by the off-microphone (with the
microphone placed at a distance from the speaker) (other than the speaker's voice) The problem
is not to pick up the sound) and (2) how to not pick up the loud sound (so that howling does not
occur). A hands-free loudspeaker system using a microphone array seeks to solve the above two
points by accurately grasping the position of the speaker and narrowing the sound collection
beam to the speaker position.
[0004]
FIG. 6 is a view schematically showing a hands-free loudspeaker system. The speech of the
speaker 51 is collected by the sound collection microphone 52, which is a microphone array,
amplified by the amplifier 53, amplified from the speaker 54, and reaches the listener 55. Here,
56 indicates the acoustic feedback component of the loud sound, and it is necessary to reduce
this acoustic feedback component 56. Reference numeral 57 denotes a sound collection beam of
the sound collection microphone 52 which is a microphone array. By narrowing the beam in this
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manner, only the voice of the speaker 51 is collected and the loud sound from the speaker 54 is
not collected. Can be
[0005]
The following are prior art documents related to the invention of this application. The invention
regarding the horn microphone which attached the microphone to the horn is described in patent
document 2 and patent document 3. FIG. Non-Patent Document 2 describes a line microphone, a
horn type sound collector, a paraboloidal sound collector, and a sound pressure gradient type
microphone. Non-Patent Document 3 describes a description on a constant directivity horn.
Japanese Patent Laid-Open No. 11-55784 Japanese Patent Laid-Open No. 2002-44772 Japanese
Patent Laid-open No. 2002-44774 Kazunori Kobayashi, "Study on application of microphone
array to the remote education system acoustic part", Japan Acoustical Society MA 2002-12, (July
26, 2002), Nakajima Heitaro, Real teaching science and technology whole book audio
engineering , real teaching publishing corporation, June 10, 1973, p. 203-205, 210-215 DB
KEELE, JR. "WHAT'S SO SACRED ABOUT EXPONENTIAL HORNS? AES Preprint # 1038, presented
at the 51st AES Convention, May 13-16, 1975
[0006]
As mentioned above, it has been proposed to implement a hands-free loudspeaker system by
focusing the beam on the speaker using a microphone array. However, in a loudspeaker system
using a microphone array, a sound collecting beam forming unit for forming a thin beam and a
speaker position detecting unit for scanning the beam to detect a sound source position are
required. Further, in the microphone array, it is necessary to arrange a plurality of microphones
at a pitch of about λ / 4, and in order to obtain a constant directivity characteristic from low to
high frequencies, the microphone interval is set for each band and the microphones are set In
addition, it is necessary to set and control delay and level individually for each microphone. As a
result, the size of the system and the size of the signal processing block become large. Also, in
terms of maintenance, the load is large because the number of microphones used is large and the
signal processing unit is present. Furthermore, there is a problem that the individual
microphones are also inferior in terms of robustness because they change with time.
Furthermore, there are also variations in the characteristics among a plurality of microphones, so
that adjustment is troublesome, and in terms of sound quality, signal processing is performed,
and there is a problem that deterioration is large.
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[0007]
As described in Non-Patent Document 2 above, as microphones having superdirectivity such as
microphone arrays, sound collectors and line microphones (gun microphones) are known, and it
is conceivable to use them. . However, gun microphones (line type) have the same disadvantages
as the above microphone array, and are very expensive. There is also a single microphone with
high directivity, but there is a problem in price. When using a reflector type (parabola type or
mirror type), complex signal processing is not necessary and systematization is easy, but in order
to lower the lower limit frequency, the diameter of the parabola or mirror is required. The focus
position of the parabola depends on the frequency. In the case of parabola, a frequency of 900
mm in diameter and taking a focal length close to the theoretical value is 2.4 kHz, and there is a
problem that even for voice, the effect can not be obtained for several hundreds Hz.
[0008]
By the way, in the case of a loudspeaker system in a conference, a seminar, a remote conference
(class), etc., it is sufficient if the voice of the speaker present in a certain area can be clearly
picked up and the other sounds can not get around. Therefore, the present inventors have used a
plurality of microphones distributed in the ceiling to pick up a sound, and the speaker specified
as a sound source position is a hand that is made to louden a signal picked up and output from a
speaker We propose a free speaker system (Japanese Patent Application No. 2005-19214). In
the proposed application of the hands-free loudspeaker system (off-microphone loudspeaker
system), the directivity characteristic of the sound pickup system is not superdirectivity required
for the general sound pickup microphone described above, but a sound pickup cover area It is to
obtain the voice of the speaker present inside, and it is not required to narrow the collection
beam to one speaker unnecessarily. That is, it may be a directional characteristic having a cover
area as indicated by 58 in FIG.
[0009]
In such a hands free loudspeaker system, when the distance between the sound source (speaker)
and the microphone becomes large, it is impossible to pick up the sound clearly. In order to avoid
this, the desired sound (voice of an unspecified speaker in a predetermined area) is efficiently
collected, and the other sounds (other reverberation, reflected sound, environmental sound, etc.)
are collected. Also, it is required that sound be picked up without lowering the degree of clarity
even with an off microphone. Therefore, the present invention provides a hands-free loud-
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speaking system which is hard to generate howling and is capable of loud and clear sounding
without requiring a complicated configuration and signal processing, and a sound collector
suitable for use in the loud-speaking system. The purpose is to
[0010]
In order to achieve the above object, the loudspeaker system of the present invention is a
plurality of sound collectors distributed on a ceiling or a wall, each sound collector being
attached to an acoustic horn and a throat portion of the acoustic horn A sound source position is
detected based on input signals from a plurality of sound collecting devices, each having a
microphone and collecting sounds in a predetermined area, and the plurality of sound collecting
devices, and detected as a sound source position. Sound source position detecting means for
selecting an input signal from the collected sound collecting device, amplifying means for
amplifying the input signal from the sound collecting device selected by the sound source
position detecting means, and the signal amplified by the amplifying means And an output
speaker. Further, the sound collecting apparatus further includes a plurality of microphones
disposed at predetermined intervals in the opening of the acoustic horn, and outputs from the
microphones attached to the throat of the acoustic horn are input and have a predetermined
frequency or more. A filter that passes a high frequency band, a filter that receives the sum of
outputs from a plurality of microphones disposed at the opening of the acoustic horn, and passes
a low frequency band below the predetermined frequency, and a filter that passes the high
frequency band An adder for adding an output and an output of the filter passing the low band
and outputting the added signal, and the directivity characteristic of the acoustic horn and the
directivity characteristic of the plurality of microphones are substantially equal at the
predetermined frequency. It is a thing. Furthermore, according to the sound collection device of
the present invention, an acoustic horn, a microphone attached to a throat portion of the acoustic
horn, a plurality of microphones disposed at predetermined intervals in an opening of the
acoustic horn, and the acoustic horn An output from a microphone attached to the throat portion
is input, and a filter passing a high frequency band above a predetermined frequency, and an
output from a plurality of microphones disposed in the opening of the acoustic horn are input,
and a low frequency below the predetermined frequency And an adder for adding the output of
the filter passing through the high band and the output of the filter passing through the low
band, and outputting the directivity characteristics of the acoustic horn at the predetermined
frequency The directional characteristics of the plurality of microphones are substantially the
same.
[0011]
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According to the loudspeaker system and the sound collector of the present invention, the sound
in the cover area can be efficiently collected and the sound outside the cover area can be
reduced, so the ratio of direct sound to indirect sound can be reduced. High sound collection is
possible. Then, the acoustic feedback component of the loud sound can be reduced, and since the
positional relationship between the microphone and the speaker does not change, it is possible to
construct a system in which howling does not easily occur. Therefore, it is possible to raise the
level of the loud sound, and it is possible to perform a loud sound with a high degree of clarity.
Further, unlike the array type and the line type, since the microphone can be used alone, stable
characteristics can be obtained. Furthermore, since it does not require a complicated
configuration or a signal processing unit, it can be configured inexpensively. Furthermore, in
addition to the horn, in the case of using a plurality of microphones arranged at predetermined
intervals in combination, it is possible to obtain uniform directional characteristics from the low
band to the high band in the voice band.
[0012]
FIG. 1 is a block diagram showing an entire configuration of an embodiment of a loudspeaker
system according to the present invention. In this figure, 1 is a plurality of (m) sound collectors
(MIC 1 to MICm) distributed on the ceiling of a room (meeting room or hall) in which the
loudspeaker system of the present invention is installed. A plurality of (n) speakers (SP1 to SPn)
are arranged. Here, in this embodiment, each sound collector 1 (MIC1 to MICm) is referred to as
an acoustic horn (hereinafter simply referred to as "horn"). And a microphone attached to the
throat thereof (hereinafter simply referred to as "horn microphone"). And the opening of the horn
is provided on the ceiling surface.
[0013]
The directivity characteristics of each horn microphone MIC1 to MICm are controlled so as to
collect only the sound generated in each predetermined cover area, and m horn microphones 1
distributed on the ceiling all It is intended to cover the room (or the area where the speaker may
be present). As the shape of the horn, various types such as an exponential horn, a conical horn,
and a parabolic horn can be used, but the directivity characteristics become constant regardless
of the frequency as described in Non-Patent Document 3 above. It is desirable to use a constant
directional horn that is designed to For example, if the distance from the speaker's mouth (sound
source) is about 1.5 m to 2 m (3 to 4 m in ceiling height), the voice of the speaker present in a 3
m square (□ 3 m) cover area is Clearly, sound can be collected (RaSTI (Rapid Speech
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Transmission Index) value of 0.8 or more, D50 of 80% or more).
[0014]
FIG. 2 is a view for explaining a cover area of the horn microphone. In this figure, 11 is a ceiling,
12 is a first horn microphone, 13 is a horn of the first horn microphone 12, 14 is a microphone
attached to the throat of the horn 13, and 15 is adjacent to the first horn microphone A second
horn microphone 16 is a horn of the second horn microphone 15, and a microphone 17 is
attached to the throat of the horn 16. The coverage area of each horn microphone is determined
from the directivity characteristics of each horn and the distance between the speaker (sound
source). As shown, the horn microphone 12 collects the sound in the cover area (1), and the horn
microphone 15 collects the sound in the cover area (2). At the height of the speaker's mouth, the
cover area of each horn microphone is set to cover the room without a gap. In addition, if the
opening size of the horn is about 600 mm, appropriate directivity characteristics can be obtained
at a frequency of 500 Hz or more.
[0015]
The plurality of speakers 5 (SP1 to SPn) are also considered to have a directivity characteristic
limited so as to louden only in the vicinity area of each, and cover the entire room with n
speakers 5 dispersedly arranged on the ceiling It is made to be able to. The arrangement interval
of the plurality of horn microphones 1 and the arrangement interval of the plurality of speakers
5 are determined according to the directivity characteristics and the ceiling height. However, it is
desirable to arrange the horn microphone and the speaker as far apart as possible. Further, the
speakers 5 do not have to be configured to be dispersedly arranged in a plurality on the ceiling,
and may be a speaker disposed on the front of the room and a speaker disposed on a wall surface
or a ceiling on the rear. Furthermore, a plurality of horn microphones may be dispersedly
disposed on the wall of the room instead of the ceiling, and a desired area may be covered.
[0016]
2 monitors the level of the input signal from each horn microphone (MIC1 to MICm) of the
plurality of horn microphones 1 to detect the position of the speaker, and controls the input
switching unit 3 and the speaker output adjustment unit 4 A sound source position detecting unit
for outputting a signal, 3 is an input switching unit for selecting a signal inputted from a horn
microphone MICi corresponding to the position of the speaker based on the control signal from
the sound source position detecting unit 2, A speaker output adjustment unit that amplifies the
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signal selected by the switching unit 3 to a level corresponding to each of the plurality of
speakers 5, and performs corresponding delay control to output the signal to the plurality of
speakers 5 (SP1 to SPn) .
[0017]
The sound source position detection unit 2 monitors the input signals from the plurality of horn
microphones (MIC1 to MICm), and among the input signals having a predetermined level or
more, the microphone MICi having the highest input signal level is used as the sound source
position (speaker position It is determined that
When the speaker stops speaking and there is no horn microphone having an input signal equal
to or higher than a predetermined level, it is determined that there is no sound source position.
Further, when the sound source position detection unit 2 outputs the input signal from the
microphone MICi determined to be the sound source position from the plurality of speakers 5
(SP1 to SPn) and outputs the signal, the sound source position detection unit 2 can The control
signal for setting the output level and the delay time (delay) for the signal output from each of
the speakers (SP1 to SPn) so that the sound pressure level at the height of the listening position
becomes uniform, the speaker output adjustment unit Output to 4.
[0018]
Here, with regard to the output signal level from each speaker, the output level of the speaker is
determined so that the sum of the direct sound from the speaker and the loud sound from the
speaker is constant at any position in the room. That is, the output level of the speaker located at
a position far from the sound source position is controlled so as to compensate for the direct
sound distance attenuation, and calculation is performed based on the distance between the
sound source position (the position of the horn microphone) and each speaker. The output level
of the speakers may be determined, or a table in which the output level corresponding to each
speaker is recorded in advance for each sound source position is created, and the output level of
the signal output from each speaker by referring to the table You may decide to Further, the
delay amount is to assign a delay time corresponding to a time required for direct sound emitted
from the sound source position to reach each speaker position to the loud sound signal outputted
from each speaker. It may be calculated based on the distance between the position of the horn
microphone and each speaker, or a table in which the delay time to each speaker is recorded in
advance for each sound source position is referred to. The amount of delay may be determined
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accordingly.
[0019]
The input switching unit 3 selects an input signal from the horn microphone based on the output
signal from the sound source position detection unit 2 (a signal specifying the horn microphone
detected to be the sound source position), and Output to the output adjustment unit 4. The
speaker output adjustment unit 4 outputs a signal to each of the plurality of speakers 5 based on
a control signal supplied from the sound source position detection unit 2 with respect to the
input signal selected by the input switching unit 3 It amplifies up to a specified output level and
gives a delay corresponding to a specified delay amount. Here, when the speaker stops speaking,
the signal specifying the sound source position is not output from the sound source position
detecting unit 2, and the input switching unit 3 does not output the input signal to the speaker
output adjusting unit 4. Then, when another speaker starts speaking, the sound source position
detection unit 2 determines that the horn microphone MICj in the vicinity of the speaker who has
newly started speaking is the sound source position, and identifies the horn microphone Output
signal to the input switching unit 3. As a result, an input signal from the horn microphone MICj is
supplied to the speaker output adjustment unit 4, and an output signal having an output level
when the horn microphone MICj is at the sound source position and subjected to a
corresponding delay is generated. Is output from each speaker 5.
[0020]
In addition, when a plurality of speakers speak at the same time and there are a plurality of
sound sources, it is possible to simultaneously carry out a plurality of systems of loud speech. In
the following, the case of performing two systems of loudspeakers will be described. Monitors
input signals from multiple horn microphones (MIC1 to MICm), and determines that the sound
source is located at these two microphones MICi and MICj when there are two microphones MICi
and MICj with input signals of a predetermined level or higher. Then, turn on MICi and MICj
(select the signal from MICi and MICj). If the speaker near the MICi stops speaking and there is
no input signal above the predetermined level in the MICi, it is determined that the sound source
at that position has disappeared, and the MICi is turned off. Furthermore, if it is determined that
the sound source has run out and another horn microphone MICk receives an input signal of a
predetermined level or more, it is determined that the sound source position has moved or a new
sound source has been generated in that microphone. Turn on MICk. When there are a plurality
of sound sources, the output of each speaker is made so that the sound pressure level becomes
uniform at any position in the room, as in the case of one system described above, for each horn
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microphone corresponding to the plurality of sound source positions. Control the level and the
amount of delay to perform a loud sound. In this case, the input switching unit 3 selects input
signals from a plurality of (for example, two) horn microphones, and the speaker output
adjustment unit 4 can process input signals of a plurality of systems. The level of the signal to be
output to each speaker and the delay amount may be controlled with respect to the input signal
of (1), and the output signals of both systems may be added and output to each speaker.
[0021]
Thus, according to the loudspeaker system of the present invention, it is possible to uniformly
collect the speaker's voice within the coverage area of the horn microphone instead of narrowing
down the sound collection beam to a specific speaker, and at the same time outside the coverage
area. Sound can be reduced and louder louder.
[0022]
In the sound collection system of the above-described loudspeaker system, the size of the mouse
(opening size) and the depth of the horn are determined by the allowable dimensions of the place
where the sound collector is attached.
Once the size of the horn is determined, the lower limit frequency that can be controlled by the
horn is determined. The following equation (1) is used to obtain a desired directivity angle only
by the configuration of a constant directivity horn + microphone (p. 18 of Non-Patent Document
3). F = K / θX (1) where: F: lower limit frequency (Hz) K: constant (25000) θ: desired directional
angle (°) X: width of horn mouse (m)
[0023]
From the above equation (1), when it is intended to obtain constant directivity at a directivity
angle of 90 ° at 500 Hz (limit frequency of the low band) or more, the horn size is as follows: X
= 25000 / (90 ° × 500 Hz) = 0.56 m. In fact, in the JBL (TM) 2352 constant directivity horn
etc., the angle that the mouse size is 559 × 457 mm and the depth is −6 dB at 254 mm is 90 °
and 50 °, and the low frequency limit frequency of 630 Hz is realized. However, below this
frequency, 90 ° and 50 ° are not satisfied. For example, in order to realize a directivity angle of
80 ° up to 200 Hz, the horn opening needs a large size of 1.6 m, and it is not realistic to arrange
it on a ceiling or the like.
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[0024]
Therefore, a second embodiment of the loudspeaker system of the present invention will be
described. In this embodiment, the directivity angle is controlled by the horn to the extent that it
is dimensionally acceptable, and a high range having constant directivity characteristics is
collected, and the low frequency region where the directivity characteristics can not be
controlled by this horn is low. By collecting the low frequency band using a microphone array in
which the directivity of the sound range is controlled, a sound collecting device capable of
obtaining uniform directional characteristics from the low frequency band to the high frequency
band is used. FIG. 3 is a diagram showing a configuration of a sound collection device according
to a second embodiment of the present invention. As shown in this figure, the sound collection
device 21 of this embodiment has a horn microphone 22 and a microphone array 26. As
described above, the horn microphone 22 is composed of the horn 23 and the microphone 24
attached to the throat thereof, and a high-pass filter (HPF) which allows a voice signal collected
by the microphone 24 to pass a signal of a predetermined frequency or more. 25 is applied to
one input of the adder 31. The microphone array 26 is composed of a plurality of microphones
27, 28 and 29 arranged at predetermined intervals, and the outputs from the microphones 27 to
29 are added together to pass a signal having a frequency equal to or lower than the
predetermined frequency. The output of the LPF 30 is supplied to the other input of the adder
31. A signal of the sum of the high frequency signal collected by the horn microphone 22 and
the low frequency signal collected by the microphone array 26 is output from the adder 31, and
this is the output of the sound collection device 21. The sound source position detection unit 2
and the input switching unit 3 in FIG. 1 are input. As described above, if the horn size of the horn
23 is 0.56 m, a 90 ° directivity angle can be obtained at 500 Hz, and as described later, the
microphone array 26 also produces a 90 ° directivity angle at 500 Hz. You can get it. By
obtaining a signal of 500 Hz or more collected by the horn microphone 22 through the HPF 25
and obtaining a signal of 500 Hz or less collected by the microphone array 26 through the LPF
30, uniform directivity from low to high regions can be obtained. It is possible to collect sounds
by characteristics.
[0025]
In general, as the usage of the microphone array, it is aimed at squeezing a beam to one sound
source and collecting only the sound as in the above-mentioned prior art, in which case the
directivity characteristic of the low band is In order to narrow down, the array length needs at
least the wavelength of the frequency. For example, to focus the sound source at 500 Hz, a
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length of 68 cm (= 34000/500) or more is required. However, as in the case of the hands-free
loudspeaker system according to the present invention, in order to collect sound in a
predetermined area with an off microphone and to reduce feedback of loudspeaker sound
(speaker output), focus on the sound source. It is sufficient to have a pointing angle with an
appropriate size, in which case the array length may be short and there is no need to control
delay time or level. Therefore, it is possible to obtain a desired directional characteristic simply
by simply adding a plurality of arranged microphone outputs.
[0026]
FIG. 4 is a polar pattern showing directivity characteristics at 500 Hz when three nondirectional
microphones are arranged at intervals of 14.5 cm. Here, three microphone outputs are simply
added without control of level and delay. According to this pattern, it is understood that the angle
satisfying −6 dB is 90 ° (45 ° on one side). Incidentally, in the case of 200 Hz, the microphone
interval is 0.34 m, and the total length of the microphone array is 0.68 m. The system ceiling is
in units of 60 cm and is of a practical size. Thus, it can be confirmed that the merits of the
microphone array come out.
[0027]
FIG. 5 is a view schematically showing an arrangement example of a horn microphone and a
microphone array in the sound collecting device 21. As shown in FIG. In this figure, 40 is a horn,
41 is a microphone attached to the throat of the horn 40, and 42 to 50 are microphones
constituting a microphone array, and are arranged in 3 rows x 3 rows in a plane at the opening
of the horn. ing. The number and arrangement of the microphones constituting the microphone
array are not limited to the example shown in this figure, and can be arbitrarily changed
according to each case. For example, only 43, 45, 46, 47 and 49 in FIG. 5 may be provided and
arranged in a cross of 1 row × 1 column. In this case, when the arrangement direction of the
listener changes, for example, in the east-west direction and the north-south direction according
to the form of the conference, the microphones in the corresponding columns or rows can be
switched and selected to collect the sound. This makes it possible to reduce the number of
microphones and simplify the apparatus.
[0028]
As described above, according to the present embodiment, it is possible to control the directivity
of the sound collection device by utilizing the merits of the horn and the microphone array
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because the purpose is not to focus on the sound source. It becomes possible to construct a
highly realistic sound collector in terms of size, shape and cost.
[0029]
It is a block diagram showing composition of a loudspeaker system of the present invention.
It is a figure for demonstrating the cover area of the sound collection apparatus in the loud
sound system of this invention. It is a block diagram which shows the structure of 2nd
Embodiment of the sound collector used for the loud-sound system of this invention. It is a figure
which shows an example of the directivity of a microphone array. It is a figure which shows the
example of arrangement ¦ positioning of each microphone in the sound collection apparatus of
this invention. It is a figure for demonstrating a hands-free loudspeaker system.
Explanation of sign
[0030]
1: sound collection device 2: sound source position detection unit 3: input switching unit 4:
speaker output adjustment unit 5: speaker 11: ceiling surface 12 15: horn microphone 13 16:
horn 14 17: Microphone, 21: Sound collector, 22: Horn microphone, 23: Horn, 24: Microphone,
25: HPF, 26: Microphone array, 27, 28, 29: Microphone, 30: LPF, 31: Adder, 40 : Horn, 41 to 50:
Microphone
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