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JP2001112083

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DESCRIPTION JP2001112083
[0001]
BACKGROUND OF THE INVENTION 1. Field of the Invention The present invention relates to a
sound image localization and amplification method using a distributed speaker, and more
particularly to a technology for expanding voices and performance sounds in a space used for
lectures, ceremonies, meetings and other meetings. .
[0002]
2. Description of the Related Art FIG. 16A shows a concentrated loudspeaker system which is an
example of a conventional loudspeaker system, and FIG. 16B shows a distributed arrangement
loudspeaker system which is another example. In the concentrated loudspeaker system, a
microphone is installed in the vicinity of a sound source O such as a speaker or a player, and a
sound pressure change caused by an utterance is amplified by a microphone amplifier and a
power amplifier and amplified by one or more speakers S. The sound from the sound source O is
transmitted to the listener 2. The speakers S are usually arranged on both sides of the stage.
Further, in the distributed arrangement and sound amplification method, the sound of the sound
source O is simultaneously amplified from a plurality of speakers S0 to S4 arranged at a constant
interval on a ceiling or the like. FIGS. 17A and 17B show times at which the loud sound from
each speaker reaches the sound receiving point in the concentrated sound amplification system
and the distributed sound amplification system (hereinafter referred to as loud sound arrival
time). And the sound pressure level of the loud sound reaching the sound receiving point
(hereinafter referred to as the loud sound reaching sound pressure level). An example of the
relationship with
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1
[0003]
However, there are the following problems in the conventional concentrated sound distribution
method and distributed arrangement sound generation method.
[0004]
(1) The direction to see and the direction to hear are different.
In the concentrated loudspeaker system where the speakers are on both sides of the stage, the
speaker O hears the voice of the speaker O from the right speaker when the seat position is on
the right side of the symmetry axis at the center of the stage. When on the left side, the speaker
O's voice can be heard from the left side speaker. Further, in the distributed arrangement
loudspeaker system, the voice of the speaker O on the stage can be heard from the overhead
ceiling speaker. In either method, the listener 2 has a problem that the recognition direction of
the speaker O differs between visual and auditory sense.
[0005]
(2) Sound sounds blurry. In the distributed arrangement and loudspeaker system, as shown in
FIG. 17 (B), a difference in direction and time occurs in the sound arriving from the plurality of
speakers S0 to S4 to the sound receiving point, and the interference of the sound waves from
each of the speakers S0 to S4 As a result, many peaks and valleys occur in the frequency
characteristics at the sound receiving point. As a result, the sound of the sound source is blurred
and unclear at the sound receiving point.
[0006]
(3) The loudness of the sound is unnatural. In the concentrated diffusion method, by placing the
speaker S near the center of the stage, it is possible to eliminate the mismatch between the visual
and auditory directions. However, the intensity of the sound differs greatly between the place
near the stage and the far place, and the problem is that the near area is noisy and can not be
heard in the far field. On the other hand, if all the distributed speakers are amplified at the same
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volume, a uniform sound field can be obtained, but there is a problem that the listener feels that
the appearance of the speaker and the size of the sound are unnatural. If you hear at a volume
that is the same as a close position, even at a position away from the speaker, the sound is
unnaturally felt because it is significantly different from the everyday experience, that is, the
experience that the voice becomes small when you leave the speaker. This unnaturalness is said
to produce results such as impairing the feeling (interactivity) spoken by the speaker, being
unable to concentrate on the talk, or failing to convey what the person wants to say.
[0007]
As a sound amplification method for solving the above-mentioned problems, as shown in FIG.
16C, a main sound source speaker (main speaker) S0 is disposed in the vicinity of the sound
source O, and distributed speakers S1 to S1 on the ceiling away from the sound source. There is a
semi-distributed loudspeaker system in which S3 is arranged. In this method, in order to obtain
the feeling that sound comes from the direction of the speaker (sound image localization in the
direction coincident with the speaker direction) even at the sound receiving point close to the
distributed speaker, Control to delay the An example of the relationship between the arrival time
at the sound reception point of the loud sound from each of the speakers S0 to S3 according to
the semi-dispersed sound amplification method and the reaching sound pressure level is shown
in FIG.
[0008]
The direction of sound image localization is based on human auditory perception, referred to as
the antecedent sound effect. The preceding sound effect is, for example, as shown in FIG. 8 (B),
when a sound from the direction of the speaker A (preceding sound) is heard at the sound
receiving point, the time delay and constant range of a certain range from the speaker B in
another direction This is a psycho-acoustic effect that localizes another sound (reinforcement
sound) that arrives with a sound pressure level difference in the range in the direction of the
speaker A (direction of the preceding sound). FIG. 8 (A) shows the arrival time difference between
the preceding sound and the reinforced sound at which the preceding sound effect is obtained at
the sound receiving point (delay time, Tb-Ta = the sound receiving point arrival time of the
reinforced sound-preceding sound received) The relationship between the sound point arrival
time) and the arrival sound pressure level difference (Lb−La = the sound reception point arrival
sound pressure level of the reinforcement sound−the sound reception point arrival sound
pressure level of the preceding sound) is shown. The shaded area in the figure is the area where
the preceding sound effect can be obtained, that is, the sound image localization area. The
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preceding sound effect may be called the Haas effect or the law of the first wave front.
[0009]
In the semi-dispersed loudspeaker system, as shown in FIG. 17C, the sound from the main
speaker S0 is controlled to reach the listener 2 at the receiving point first. Moreover, the
reinforcement sound from dispersion | distribution speaker S1-S3 (following, it may be called
subsequent sound. ) Is controlled to reach the listener 2 at the sound receiving point with a time
delay of about 10 ms from the preceding sound from the main speaker S0 and a sound pressure
level difference of a certain range shown in FIG. 8 (A). As described above, by controlling the
arrival time and the reaching sound pressure level of the loudspeaker sound from each of the
speakers S0 and S1 to S3, it is possible to localize the sound image in the main speaker direction
at the sound receiving point.
[0010]
However, in the conventional semi-dispersed loudspeaker system, it is difficult to obtain sound
image localization at all sound receiving points in the sound field due to the difference in timbre
of the main speaker S0 and the dispersive speakers S1 to S3. There is a problem. In order to
obtain the leading sound effect, it is necessary that the timbre of the leading sound and the
subsequent sound, that is, the frequency characteristics be close. In particular, it is necessary that
there is no difference in the frequency characteristics of the preceding sound and the following
sound in the middle range (250 Hz to 1 kHz). In the semi-dispersed loudspeaker system, the
main speakers S0 and the dispersed loudspeakers S1 to S3 have different performances due to
the difference in their roles, so the loudspeakers from both loudspeakers have the same
frequency characteristics at all sound receiving points in the sound field. It is difficult to convey
in the background, and the generation of a sound receiving point where the leading sound effect
can not be obtained is inevitable.
[0011]
In addition, the semi-dispersed loudspeaker system has a problem that it can not cope with
changes in the position of the sound source. In order to obtain sound image localization, the
deviation between the direction of the speaker (sound source) viewed from the listener and the
direction of the preceding sound speaker needs to be within a certain tolerance. Referring to FIG.
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7, the speaker (sound source) and the listener (reception point) are on opposite sides (positive
side and negative side) with the origin on the Y axis on the XY plane (horizontal plane), and the
preceding sound is Assuming that the speaker is on the XZ plane, a range of ± 10 degrees in the
X-axis direction and ± 45 degrees in the Z-axis (vertical) direction with respect to the speaker
direction (Y-axis direction) viewed from the listener The hatched range) is taken as the placement
allowable range of the preceding sound speaker. When the position of the speaker changes in the
semi-distributed amplification method, for example, when the stage position changes, the main
speaker S0 must be moved within the arrangement allowable range of FIG. 7 according to the
position of the speaker. It is difficult to move the main speaker S0. Further, even when a
questioner or the like appears at an unspecified position in the customer seat, it is also
impossible to cause the questioner to localize the sound image. For this reason, there are
problems such as not being able to know the position of the questioner in a large lecture hall or
the like. It is desired to develop a method of sound amplification in which the direction of sound
image localization changes in accordance with the change in the position of the sound source and
the appearance of the questioner.
[0012]
Furthermore, in the semi-dispersed loudspeaker system, there is also a problem that the
difference between the transmission characteristic from the main speaker S0 and the
transmission characteristic from the distributed speakers S1 to S3 for both ears of the listener 2
deviates from the allowable range. In order to obtain the preceding sound effect, it is necessary
that the transmission characteristics of the preceding sound speaker and the subsequent sound
speaker for both ears of the listener 2 are similar. That is, the direction of the preceding sound
speaker and the direction of the subsequent sound speaker as viewed from the sound receiving
point must be within a predetermined allowable range. The preceding sound speaker needs to be
within the range shown by the oblique lines in FIG. 7, that is, within ± 10 degrees in the
horizontal direction and ± 45 degrees in the vertical direction with respect to the sound source
direction from the sound receiving point. Preferably, the subsequent sound speaker is in a cone
with an apex angle of 90 degrees with the central axis in the direction immediately above
(vertical) the sound receiving point. In the semi-dispersed loudspeaker system, it is difficult to
arrange the main speaker S0 and the dispersed speakers S1 to S3 so that the transmission
characteristics for the listeners' ears are within the allowable range, and a sound receiving point
where the preceding sound effect can not be obtained occurs. There is a case.
[0013]
Therefore, an object of the present invention is to provide a sound image localization and sound
amplification method and a sound amplification system using distributed speakers which can
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change the direction of sound image localization in accordance with the direction of a sound
source.
[0014]
DETAILED DESCRIPTION OF THE PREFERRED EMBODIMENT Referring to the embodiment of
FIG. 1, the sound image localization and sound expansion method using distributed speakers
according to the present invention is substantially used for the sound amplification of the sound
source O at a predetermined position above the sound receiving surface Ph. A plurality of
speakers Sj (1 ≦ j ≦ n) having the same acoustic characteristics are distributed downward by
partially overlapping the cover areas of the adjacent speakers; a portion on the sound reception
surface Ph vertically below each speaker Sj Among the speakers Sj, the speaker closest to the
sound source O is selected as the main speaker So, and the remaining speakers are ranked in
ascending order of the distance from the main speaker So. Speaker Sgx (x = 1, 2,..., (N-1)); The
loud sound of the sound source O from the main speaker So is produced at the time zero without
time delay with the required input power; The peripheral speakers Sgx (in the ascending order of
the ranks for each) For example, Sg15) At the lower sound receiving point Pgx (for example,
Pg15), the loud sound arrival time Txx from the peripheral speaker Sgx is delayed by the delay
time Δtx that gives the preceding sound effect to the loud sound arrival time Tox from the main
speaker So (Txx The sound generation time of the peripheral speaker Sgx (for example, Sg15)
such as = Tox + Δtx is calculated, and the sound reception point Pgx (for example, Pg15) is
reached prior to the loud sound arrival time Txx from the peripheral speaker Sgx (for example,
Sg15) The sum of the reaching sound pressure levels of the lower order peripheral speakers Sgk
(k = 1, 2,..., (X-1)) and the loud sound from the main speaker So is received from the respective
speakers Sgk and So Obtained as a function of the distance to Pgx and the input power and
acoustic characteristics of each speaker Sgk and So and at the sound reception point Pgx (for
example Pg15) the loud sound arrival sound pressure level Lxx from the relevant peripheral
speaker Sgx (for example Sg15) To the sum calculated above Sound pressure level difference
ΔLx which gives the preceding sound effect higher, and the input power of the peripheral
speaker Sgx (for example, Sg15) corresponding to the determined arrival sound pressure level
Lxx is calculated; from each of the peripheral speakers Sgx By sounding the loud sound of the
sound of the sound source O at the calculated sounding time with the calculated input power, the
localization direction of the sound image at each sound receiving point Pi is made to be the
direction toward the sound source O.
[0015]
Preferably, a predetermined target sound pressure level Lhc which is the upper limit of the
listening sound pressure level at each sound receiving point Pi is provided, and the listening
sound pressure level at the sound receiving point Pgx (for example Pg15) below each peripheral
speaker Sgx (for example Sg15) The loud sound arrival sound pressure level Lxx from the
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peripheral speaker Sgx and the low priority peripheral speakers Sgk (k = 1, 2,..., (X-1)) and the
loud sound from the main speaker So Obtained as a total with the sum of the reached sound
pressure levels, and when the total exceeds the target sound pressure level Lhc, the loud sound
reaching sound pressure level Lxx of each peripheral speaker Pgx (for example, Sg15) is within
the range where the preceding sound effect is obtained. The total amount is controlled to
coincide with the target sound pressure level Lhc.
[0016]
More preferably, as shown in FIG. 5, an inclined target in which the predetermined target sound
pressure level Lhc is reduced according to the distance from the sound source O, and the
reduction slope with respect to the distance is smaller than the attenuation slope in the free
sound field. The sound pressure level Lhc is used as the sound pressure level, and the listening
sound pressure level at each sound receiving point Pi on the sound receiving surface Ph is gently
reduced according to the distance from the sound source O.
[0017]
Further, referring to the embodiment of FIG. 1, the sound image localization and sound
amplification system using distributed speakers according to the present invention is
substantially arranged such that the cover areas of the adjacent speakers are partially overlapped
and distributed downward at predetermined positions above the sound receiving surface Ph.
Upper speaker Sj (1 ≦ j ≦ n) for the same acoustic characteristic; signal transmission device 10
for transmitting an acoustic signal from the sound source O to each speaker Sj; provided between
the signal transmission device 10 and each speaker Sj And the signal adjustment device 20 which
adjusts the sounding time and input power of the acoustic signal for each speaker Sj according to
the instruction input and outputs it to each speaker Sj; the speaker closest to the sound source O
among the speakers Sj is the main speaker So Speaker selection means 31 to select; speakers Sj
are ranked in ascending order of distance from the main speaker So as speakers Sj other than the
main speaker So as peripheral speakers Sgx (x = 1, 2,..., (N-1)) Means 32 Main speaker sound
instruction means 33 for instructing the sounding time and input power of the main speaker So;
and a sound receiving point Pi (1 ≦ i ≦ n) on the sound receiving surface Ph vertically below
each speaker Sj, and peripheral speakers Sgx The loud sound arrival time Txx from the peripheral
speaker Sgx is the loud sound arrival time Tox from the main speaker So at the sound reception
point Pgx (for example Pg15) below the peripheral speaker Sgx (for example, Sg15) in ascending
order of the ranks. The sound generation time of the peripheral speaker Sgx (for example, Sg15)
such as Tx = Tox + Δtx, which is delayed by a delay time Δtx giving the preceding sound effect,
is calculated, and the peripheral speaker Sgx (for example, Sg15) Sum of loud sound reaching
sound pressure levels from the lower order peripheral speakers Sgk (k = 1, 2,..., (X−1)) and main
speakers So arriving earlier than the loud sound reaching time Txx from From the speakers Sgk
and So Obtained as a function of the distance to the point Pgx and the input power and acoustic
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characteristics of each of the speakers Sgk and So and at the sound receiving point Pgx (for
example Pg15), the loud sound reaching sound pressure level from the relevant peripheral
speaker Sgx (for example Sg15) The Lxx is determined to be higher by the sound pressure level
difference ΔLx giving the preceding sound effect than the total calculated above, and the input
power of the peripheral speaker Sgx (for example, Sg15) corresponding to the determined arrival
sound pressure level Lxx is calculated. The peripheral speaker sound instructing means 34 for
instructing the calculated sounding time and input power, and inputting the sounding time and
input power instructions from the main speaker sound instructing means 33 and the peripheral
speaker sound instructing means 34 to the signal adjustment device 20 Thus, the localization
direction of the sound image at each sound receiving point Pi is the direction toward the sound
source O.
[0018]
DESCRIPTION OF THE PREFERRED EMBODIMENTS The embodiment of FIG. 1 shows an
embodiment in which twenty-five loudspeakers Sj (1.ltoreq.j.ltoreq.25) are dispersed downward
above the sound receiving surface Ph.
The sound receiving surface Ph is, for example, a virtual surface assumed at the height h of the
ear of the listener 2 on the floor surface 6.
Although the figure shows a horizontal sound receiving surface Ph parallel to the floor surface 6,
the sound receiving surface Ph targeted by the present invention is not limited to the horizontal
surface.
It is desirable from the viewpoint of control of signal adjustment to be described later that the
speakers Sj be disposed downward from the sound receiving surface Ph above the fixed height so
that the mutual distance becomes constant.
[0019]
However, the distributed arrangement of the loudspeakers Sj according to the present invention
is not necessarily limited to the distributed arrangement at fixed heights and intervals.
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For example, when there is a step on the ceiling, the height from the sound receiving surface Ph
to each speaker Sj is not constant, but the sound receiving surface Ph is changed by changing the
input power of each speaker Sj according to the difference in height. It is possible to compensate
for the difference in sound pressure of the loud sound above.
Further, as described later, if the overlap of the cover area of the adjacent speakers Sj can be
obtained, for example, about 20% or more, the intervals between the speakers Sj do not have to
be constant.
[0020]
In FIG. 1, a plurality of speakers Sj having substantially the same acoustic characteristics are
used, and they are arranged at a height H where a part of the cover area of the adjacent speakers,
for example, about 20%, overlaps on the sound receiving surface Ph. Instead of using a main
speaker different from the distributed speakers as in the semi-dispersed loudspeaker system, for
example, by using the speakers Sj having substantially the same acoustic characteristics such as
sound pressure frequency characteristics and directivity coefficient, the timbre from each
speaker Sj To avoid the hindrance of the preceding sound effect due to the difference in timbre
on the sound receiving surface Ph. Further, for example, by partially overlapping a cover area of
a band of one octave band centered on 1 kHz, preferably 500 Hz to 2 kHz, generation of an area
where the sound of the band does not reach is avoided on the sound receiving surface Ph. It is
possible to obtain the leading sound effect at any position on the sound receiving surface Ph.
[0021]
In general, the characteristic A0 (f) of the speaker Sj is at a point of 1 m on the speaker front axis
(horizontal directional direction 0 °, vertical directional direction 0 °) at the time of 1 (watt)
signal input at frequency f (Hz) It is defined as a square sound pressure P2 (a unit is a square of
Pa (pascal), that is, Pa2). The sound pressure level L is defined as a squared sound pressure P2
based on a sound pressure P0 (Pa) of 20 μPa, as shown by the following equation (1). In the
following description, a squared sound pressure P2 (P2 / P02) based on a sound pressure P0 of
20 μPa is referred to as a squared sound pressure I.
[0022]
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The sound pressure frequency characteristic of the speaker is defined as the frequency
characteristic of the sound pressure level L at a point 1 m in front of the speaker at the time of 1
(watt) signal input. Also, if the sound pressure in the horizontal direction φ and the vertical
direction θ is represented as A (f, θ, φ) and A0 (f) is represented as A (f, 0, 0), the directivity
coefficient D (f, θ) , Φ) are defined as the following equation (2). The squared sound pressure I
and the sound pressure level L of the loudspeaker sound reaching the sound receiving point
separated by a distance R (m) in the direction of the horizontal pointing direction φ and the
vertical pointing direction θ with respect to the speaker are 4), the distance R (m) from the
speaker S to the sound receiving point, the directivity coefficient D (f, θ, φ) of the speaker, the
sound pressure frequency characteristic A (f) of the speaker, and the input It is expressed as a
function of power W (watt).
[0023]
Sound pressure level L = 10 log (P2 / P02) (dB) = 10 log I (dB) (1) Directing coefficient D (f, θ,
φ) = A (f, θ, φ) / A (f, 0, 0) ......... (2) Squared sound pressure of the sound receiving point P = W ·
D (f, θ, φ) · A (f) / R2 (dB) ) ...... (3) Sound pressure level of the sound receiving point P L = 10 ·
log {W · D (f, θ, φ) · A (f) / R2} (dB) (4)
[0024]
In FIG. 1, only the loudspeakers Sj having the same acoustic characteristics such as sound
pressure frequency characteristics and directivity coefficient are used.
The graph of FIG. 9 (B) shows directional frequency characteristics of an opening angle
(directional angle) at which the sound pressure level L attenuates by -6 dB with respect to the
front axis. For example, when the loudspeaker of the directional frequency characteristic of FIG.
9B is selected as the distributed arrangement loudspeaker Sj of the present invention, since the
directional angle of 1 kHz is about 85 to 90 degrees, the directional angle and the overlapping
ratio of the cover area are The arrangement height from the sound receiving surface Ph of each
speaker Sj can be designed in consideration. The speaker arrangement of FIG. 1 is an example of
the case where the speakers of the directional frequency characteristic shown in FIG. 9 are
dispersedly arranged.
[0025]
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10
However, the speakers Sj used in the present invention are not necessarily limited to the same
type. For example, when the difference in sound quality of one octave band centered on 1 kHz,
preferably 500 Hz to 2 kHz, is small and the acoustic characteristics are substantially the same,
different types of speakers can be used.
[0026]
The mutual spacing of the dispersively arranged speakers Sj is determined so that the
transmission characteristics of the two speakers Sj in the direction of the sound source O with
respect to the sound receiving point are similar. As described later, in the present invention, the
speaker Sj closest to the sound source O is used as a preceding sound speaker, and the sound
image is localized using the surrounding speaker Sj in the direction from the sound source O
toward the sound receiving point (listener 2) as a subsequent sound speaker. By appropriately
designing the distance between the distributed speakers Sj in consideration of the relationship
with the arrangement height, the sound receiving direction for the front sound speaker and the
subsequent sound speaker can be determined at an arbitrary sound receiving point on the sound
receiving surface Ph. The difference in transmission characteristics can be made within an
allowable range in which sound image localization is possible.
[0027]
Furthermore, in the above-described arrangement of the speakers Sj, the deviation between the
direction of the sound source O viewed from the sound reception point and the direction of the
preceding sound speaker is within the allowable range shown in FIG. For example, in FIG. 1, when
viewed from the listener 2 below the speaker S5, even if the speaker O is below any other
speaker Sj, the speaker Sj above the speaker with respect to the direction of the speaker O in FIG.
Since it arrange | positions in the tolerance | permissible_range, the sound image localization can
be obtained by making the speaker above a speaker into a prior sound speaker. That is, in the
present invention, the plurality of speakers Sj are distributed at predetermined positions
satisfying the heights at which the cover areas are superimposed, the mutual intervals at which
similar transmission characteristics are obtained, and the arrangement allowable range of the
preceding sound speakers.
[0028]
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11
An acoustic signal from a sound source O is input by the signal transmission device 10 to the
speakers Sj arranged in a distributed manner. The signal transmission device 10 of FIG. 1 has a
microphone 3, a mixer 11 and a signal transmission path. The microphone 3 is disposed in the
vicinity of a speaker O in the sound field and other players, and collects the direct sound. The
picked-up direct sound is amplified by a microphone amplifier (not shown), and then the volume
control of the entire sound field is performed by the mixer 11 and sent to the signal adjustment
device 20 described later. The microphone volume control in the mixer 11 can be manually
controlled by the operator in consideration of the volume of the sound source O. However, the
scope of application of the present invention is not limited to the amplification of the sound of
the sound source O in the sound field, and can be applied, for example, to localize a sound image
by sound amplification from a sound source recorded in advance.
[0029]
In the present invention, a signal adjustment device (effector) 20 for adjusting the acoustic delay
and the sound pressure level of each speaker Sj is inserted between the signal transmission
device 10 and each speaker Sj. One example of the signal adjustment device 20 is, as shown in
FIGS. 12 and 13, an acoustic delay circuit 21 that adjusts the sound generation time and input
power of an acoustic signal for each speaker Sj according to the input of an instruction signal
from the computer 30 A sound pressure level control circuit (volume control circuit) 22 is
included. Each signal conditioning device 20 is connected to the computer 30, and the acoustic
delay of each signal conditioning device 20 and the adjustment of the input power are
collectively controlled by the computer 30.
[0030]
FIG. 6 shows an example of a flow chart of control of the signal conditioning device 20 by the
computer 30 in the case where sound image localization is obtained by the sound expansion of
the sound of the sound source O in the sound field. Hereinafter, the loudspeaker method of the
present invention will be described with reference to the flowchart of FIG. First, at step 601, the
position of the sound source O in the sound field is detected. In FIG. 1, the position of the sound
source O is detected based on the detection signal of the microphone position detection device
37. Next, at step 602, the speaker selection means 31 selects the speaker S1 closest to the sound
source O as the main speaker So. An example of the speaker selection means 31 is a computer
built-in program that calculates the distance between the sound source O and each speaker Sj
04-05-2019
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and selects the speaker S1 with the smallest distance. For detection of the position of the
microphone 3, manual input by an operator or various automatic detection techniques belonging
to the prior art can be used. However, the microphone position detection device 37 is not an
essential requirement of the present invention.
[0031]
In step 603, the distributed speakers Sj other than the main speaker So are ranked in ascending
order of the distance from the main speaker So by the speaker ranking means 32, which is a
program built into the computer 30, for example. In the present specification, the ranked
speakers Sj are referred to as peripheral speakers Sgx (x = 1, 2,..., (N−1)). For example, in FIG. 1,
the speaker S2 in the distributed arrangement is the peripheral speaker Sg1 closest to the sound
source O, the speaker S3 is the fourth nearest peripheral speaker Sg4, the speaker S4 is the
eighth closest peripheral speaker Sg8, and the speaker S5 is 15 It is ranked as the third closest
peripheral speaker Sg15.
[0032]
As shown in steps 601 to 603, in the present invention, the position of the main speaker So is not
fixed, and the dispersed speaker closest to the sound source O is used as the main speaker So,
and the peripheral speakers Sgx are ranked according to the distance from the main speaker So.
Therefore, even when the sound source O moves, the position of the main speaker So and the
order of the peripheral speakers Sgx can be changed according to the movement, and the sound
image localization on the sound receiving surface Ph can be changed.
[0033]
In step 604, the sound generation time To and the input power Wo of the main speaker So are
instructed by the main speaker sound instruction means 33 which is a program built into the
computer 30, for example.
FIG. 10 shows a method of determining the sound generation time and input power of the main
speaker So. For example, the sound signal from the signal transmission device 10 is produced by
the main speaker So without time delay, and the sound production time To is set to time zero
(ms). However, even if a time delay is provided at the sound generation time To of the main
speaker So, the sound image localization of the present invention can be obtained. Also, the input
04-05-2019
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power Wo of the main speaker So is such that the loudness attainment sound pressure level from
the main speaker So at the microphone 3 is lower than the attainment sound pressure level of
the voice from the speaker O (sound pressure level at a point 1 m apart). The level required to
prevent howling, for example, a level lower by 2 dB is determined. However, when howling does
not occur, the input power Wo of the main speaker So can be set to the required level. When
determining the input power Wo of the main speaker So, the volume of the mixer 11
corresponding to the microphone 3 can be appropriately adjusted as shown in FIG. 10, and the
attenuation of the attenuator 12 of the amplifier 12 at that time (hereinafter referred to as the
attenuator position) It is said that. ) Can be set to 0 (dB).
[0034]
About the determination method of the pronunciation time and the input power of each
peripheral speaker Sgx by the peripheral speaker sound instruction means 34 after step 605, five
speakers S1 to S5 are set between the speaker O and the listener 2 for simplification of the
explanation. It demonstrates with reference to FIG. 2 arrange | positioned. FIG. 2 shows the case
where the speaker S1 closest to the speaker O is the main speaker So, the speakers S2 to S5 are
ranked as the peripheral speakers Sg1 to Sg4, and the listener 2 is below the peripheral speaker
Sg4. In step 605, first, sound receiving points P1 to P5 are determined on the sound receiving
surface Ph vertically below the speakers S1 to S5. In FIG. 2, the sound receiving point P1 is a
sound receiving point Po vertically below the main speaker So, and the sound receiving points P2
to P5 are sound receiving points Pgx (Pg1 to Pg4) vertically below the peripheral speakers Sgx
(Sg1 to Sg4). is there.
[0035]
Assuming that the distance between the main speaker So and the sound receiving point Pgx is
Rox, the time required for the loud sound from the main speaker So to reach the sound receiving
point Pgx below the peripheral speaker Sgx is as the following equation (10) It can be calculated.
(10)In the equation, the symbol C represents the velocity of sound (m / s). Usually, the sound
velocity C at normal temperature and normal humidity (23 ° C., relative humidity 60%) is used
as the sound velocity C. Also, the sound pressure level Lox of the loud sound from the main
speaker So that has reached the sound reception point Pgx can be expressed as the following
equation (11) with reference to the equation (4). (11)In the equation, Wo represents the input
power of the main speaker So, and Do represents the directivity coefficient of the main speaker
So.
04-05-2019
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[0036]
Also, the time required for reaching the loud sound from the peripheral speaker Sgx to the sound
receiving point Pgx below and the reaching sound pressure level Lxx are also main if the distance
between the peripheral speaker Sgx and the sound receiving point Pgx is Rxx. Similar to the case
of the speaker So, the equations (12) and (13) can be used. (13)In the equation, Wx indicates the
input power of the peripheral speaker Sgx, and Dx indicates the directivity coefficient of the
peripheral speaker Sgx. In the illustrated example, since the main speaker So and the peripheral
speaker Sgx have the same acoustic characteristics, A (f) in the equations (11) and (13) is the
same.
[0037]
The loud sound from the main speaker and a plurality of peripheral speakers can reach each
sound reception point Pgx. The synthetic sound pressure level (hereinafter referred to as the
listening sound pressure level) at the sound receiving point Pgx to which the loud sound from the
speakers Sj (j = 1, 2, 3,...) Of the same characteristic A (f) arrives. Can be expressed by the
following equation (14) based on the sum of squared sound pressure I of the reaching sound
pressure from each speaker Sj.
[0038]
The time required for the loud sound arrival from the main speaker So to the sound receiving
point Pgx = Rox / C (s) = 1000 · Rox / C (ms) ...................................... (10) Main Loud sound arrival
sound pressure level from the speaker So to the sound receiving point Pgx Lox = 10 log Iox = 10 ·
log {Wo · Do (fo, θo, φo) · A (f) / Rox2} (dB) (dB) 11) Time required to reach the loud sound
from the peripheral speaker Sgx to the sound receiving point Pgx = Rxx / C (s) = 1000 · Rxx / C
(ms) ...................................... (12 ) Sound pressure level Lxx = 10 log Ixx = 10 · log {Wx · Dx (fx, θx,
φx) · A (f) / Rxx2} (dB) ... about the loud sound arrival sound pressure level from the peripheral
speaker Sgx to the sound receiving point Pgx ... (13) Listening sound reception level at the sound
receiving point Pgx = 10 log (Σ Ixx) = 10 · log [Σ {W j · D j (f j, θ j, φ j) · A (f) / R k j 2}] (dB)
(14)
[0039]
04-05-2019
15
In steps 605 to 612, the tone generation time and input power of each of the peripheral speakers
Sgx are determined in ascending order of the order of the peripheral speakers Sgx.
First, for the peripheral speaker Sg1 in FIG. 2, in step 605, the distance Ro1 from the main
speaker So to the sound receiving point Pg1 below the peripheral speaker Sg1 and the distance
R11 from the peripheral speaker Sg1 to the sound receiving point Pg1 are calculated. In the case
of the peripheral speaker Sg1, there is no lower-order peripheral speaker Sgx, that is, the
peripheral speaker Sgx closer to the main speaker So than the peripheral speaker Sg1. In step
606, from the main speaker So at the sound receiving point Pg1, from the substitution result of
the distance Ro1 from the main speaker So to the sound receiving point Pg1 into the equation
(10) and the tone generation time To = 0 (ms) of the main speaker So The loud sound arrival time
To1 of is calculated. Also, based on the substitution result of the distance Ro1 from the main
speaker So to the sound receiving point Pg1 and the input power Wo of the main speaker So and
the directivity coefficient Do into the equation (11), the loud sound from the main speaker So at
the sound receiving point Pg1 The arrival sound pressure level Lo1 is calculated. However, the
sound pressure frequency characteristic A (f) of the main speaker So is predetermined.
[0040]
In step 607, a delay time .DELTA.t1 for giving a preceding sound effect to the loud sound arrival
time To1 from the main speaker So is obtained. The delay time .DELTA.t1 is the shortest delay
time .DELTA.tmin, for example 3 ms, for the peripheral speaker Sg1 closest to the main speaker
So, and the longest delay time .DELTA.tmax, for example 10 ms, for the peripheral speaker Sg4
farthest from the main speaker So The peripheral speakers Sg2 and Sg3 can be defined as times
proportionally distributed to the distances Ro2 and Ro3 from the main speaker So to the sound
receiving points Pg2 and Pg3 below the peripheral speakers Sg2 and Sg3.
[0041]
However, the method of determining the delay time Δt1 is not limited to the method by
proportional distribution according to the distance. For example, as shown in FIG. 11, a distance
Ro2 from the main speaker So to the peripheral speakers Sg2 and Pg2 below the peripheral
speakers Sg2 and Sg3, and a distance Ro2 between the shortest delay time and the longest delay
time proportional to a logarithm (logRox) of Ro3. It can be defined as the time of day. Table 1
below shows delay times Δtx of the peripheral speakers Sg1 to Sg4 determined from the graph
of FIG. However, in Table 1, the delay time Δt1 of the peripheral speaker Sg1 which is the
04-05-2019
16
shortest delay time Δtmin is 6 ms, and the delay time Δt4 of the peripheral speaker Sg4 which
is the longest delay time Δtmax is 30 ms.
[0042]
At step 607, the arrival time T11 (= To1 + Δt1) at which the loud sound from the peripheral
speaker Sg1 should reach the sound receiving point Pg1 is calculated as the sum of the delay
time Δt1 and the loud sound arrival time To1 from the main speaker So. Do. If the time T11 at
which the loud sound should arrive is known, the distance R11 from the peripheral speaker Sg1
to the sound receiving point Pg1 is substituted into the equation (12), whereby the peripheral
speaker Sg1 corresponding to the loud sound arrival time T11 in step 608 is obtained. The
pronunciation time can be determined.
[0044]
In step 609, the sum of the loudness sound reaching sound pressure levels from the lower order
peripheral speakers Sgk and the main speakers So that arrive at the sound receiving point Pg1
earlier than the loudness sound arrival time T11 from the peripheral speaker Sg1 is calculated. .
In this case, the loud sound reached prior to the arrival time of the loud sound from the
peripheral speaker Sg1 at the sound receiving point Pg1 is only the loud sound from the main
speaker So, so the arrival sound pressure of the loud sound from the main speaker So Assuming
that the level Lo1 is the total of the arrival sound pressure levels of the preceding loud sound, in
step 610, a sound pressure level difference ΔL1 that gives the preceding sound effect to the
loud sound arrival sound pressure level Lo1 from the main speaker So is calculated. This sound
pressure level difference ΔL 1 is, for example, as shown in FIG. 8A, the delay time Δt from the
preceding sound necessary to give the preceding sound effect to the following sound and the
sound pressure level difference ΔL between the preceding sound and the following sound The
equation (ΔL = f (Δt)) with the above is determined in advance by experiments etc., and the
delay time Δt1 required for calculation of the sounding time of the peripheral speaker Sg1 is
substituted into the relation of FIG. 8 (ΔL1 = f (Δt1)) It can be determined by
[0045]
Based on the determined sound pressure level difference ΔL1 and the loud sound arrival sound
pressure level Lo1 (= 10 log Io1) from the main speaker So, the sound pressure level L11 that the
loud sound from the peripheral speaker Sg1 should show at the sound receiving point Pg1 in
step 612 The input power W1 of the peripheral speaker Sg1 is calculated in step 612 by
04-05-2019
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calculating = 10logIo1 + ΔL1) and substituting the calculated ultimate sound pressure level L11
and the distance R11 from the peripheral speaker Sg1 to the sound receiving point Pg1 into
equation (13). decide.
[0046]
For example, in FIG. 8, assuming that the delay time Δt1 of the arrival time of the peripheral
speaker Sg1 is 6 ms, the sound image if the difference between the arrival sound pressure level
from the peripheral speaker Sg1 and the arrival sound pressure level from the main speaker So is
within 6 dB. Indicates that localization is obtained.
The arrival sound pressure level Lo1 at the sound receiving point Pg1 of the main speaker So is
lowered due to distance attenuation or the like as shown in the equation (11). Based on the
arrival sound pressure level Lo1 at the sound receiving point Pg1 of the main speaker So, the
relative input level of the peripheral speaker Sg1 is set so that the sound pressure level from the
peripheral speaker Sg1 becomes (Lo1 + 6) (dB) at the sound receiving point Pg1. The result
determined by the attenuator position of the amplifier 12 is -3 (dB) shown in Table 1. Thus, the
attenuator position of the amplifier 12 can be adjusted or corrected so that the input power of
the peripheral speaker Sgx becomes the value calculated in step 611.
[0047]
After the tone generation time and input power of the peripheral speaker Sg1 are determined in
step 612, the process proceeds to step 613 to determine whether the calculation of the tone
generation time and input power has been completed for all the peripheral speakers. In this case,
since the processes of the peripheral speakers Sg2 to Sg4 remain, the process returns to step
605, and by repeating steps 605 to 612, the sound generation time and input power of the
peripheral speaker Sg2 are determined. In the case of the peripheral speaker Sg2, since there is a
lower order peripheral speaker Sg1, in addition to the calculation of the distance Ro2 from the
main speaker So to the sound receiving point Pg2 and the distance R22 from the peripheral
speaker Sg2 to the sound receiving point Pg2 in step 605 The distance R12 from the lower order
peripheral speaker Sg1 to the sound reception point Pg2 is calculated. Further, at step 606, the
time when the loud sound from the lower order peripheral speaker Sg1 reaches the sound
receiving point Pg2 and the loud sound reaching sound pressure level L12 are calculated based
on the equations (12) and (13).
04-05-2019
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[0048]
In step 607, a delay time .DELTA.t2 for giving a preceding sound effect to the loud sound arrival
time To2 from the main speaker So is determined, but this calculation method is the same as the
method described above for the peripheral speaker Sg1. From the sum of the delay time Δt2 and
the loud sound arrival time To2 from the main speaker So, a time T22 (= To2 + Δt2) at which the
loud sound from the peripheral speaker Sg2 should arrive at the sound reception point Pg2 is
determined. If the time T22 at which the loud sound should reach is known, the tone generation
time of the peripheral speaker Sg2 can be determined in step 608 by substituting the distance
R22 from the peripheral speaker Sg2 to the sound receiving point Pg2 in equation (12).
[0049]
In step 609, the loud sound arrival time T22 from the peripheral speaker Sg2 is compared with
the loud sound arrival time T12 from the peripheral speaker Sg1 at the sound reception point
Pg2, and if the loud sound arrival time T12 from the peripheral speaker Sg1 precedes, The sum
(I12 + Io2) of the squared sound pressure I12 of the reaching sound from the peripheral speaker
Sg1 at the sound receiving point Pg2 and the squared sound pressure Io2 of the reaching sound
from the main speaker So is determined, and the sum of the squared sound pressure (I12 + Io2
(14) to calculate the sum (= 10 log (I12 + Io2)) of the loud sound arrival sound pressure level
L12 from the peripheral speaker Sg1 and the loud sound arrival sound pressure level Lox from
the main speaker So .
[0050]
The inventors of the present invention have made it possible for the loud sound from the
peripheral speakers Sgk (k = 1, 2,..., (X−1)) of lower rank to precede the loud sound from the
peripheral speakers Sgx at each sound receiving point Pgx. If it has reached, not only the loud
sound arrival sound pressure level Lox from the main speaker So, but also the loud sound arrival
sound pressure level Lkx (= 10 log I Ikx) from the lower order peripheral speaker Sgk and the
loud sound from the main speaker So The loudness attainment sound pressure level Lxx from the
peripheral speaker Sgx above the sound reception point Pgx is given so as to give the preceding
sound effect to the sum (= 10 log (k Ikx + Iox)) with the attainment sound pressure level Lox (=
10 log Iox). It has been found that, by setting, in the wide sound receiving surface Ph, the
preceding sound effect can be obtained reliably at every sound receiving point thereon.
04-05-2019
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The present invention is based on such findings.
[0051]
In step 610, for example, the sound pressure level difference ΔL2 is determined by substituting
the delay time Δt2 of the sound generation time of the peripheral speaker Sg2 into the relational
expression of FIG. 8 (ΔL2 = f (Δt2)). The sound pressure level L22 (= 10 log (I12 + Io2) + ΔL2)
that the arrival sound from the peripheral speaker Sg2 should be indicated by the sound
reception point Pg2 is obtained by adding the level difference ΔL2 obtained to the above sum (=
10 log (I12 + Io2)). Once the reaching sound pressure level is determined, in step 612, the input
power W2 of the peripheral speaker Sg2 can be determined by substitution into the equation
(13) as in the case of the peripheral speaker Sg1 described above.
[0052]
After the tone generation time and input power of the peripheral speaker Sg2 are determined in
step 612, the process returns from step 613 to return to step 605 and repeats steps 605 to 612
for the peripheral speakers Sg3 and Sg4. By repeating steps 605 to 612 in FIG. 6 in ascending
order of rank for all the peripheral speakers Sgx, it is possible to determine the tone generation
time and input power of all the peripheral speakers Sgx. The sound production time and input
power of the acoustic signal from the signal transmission device 10 are adjusted for each speaker
Sj by inputting to the signal adjustment device 20 the sound production time and input power
instruction signals of the respective peripheral speakers Sgx determined by the computer 30
Then, the adjusted acoustic signal is input to each speaker Sj via the power amplifier 12 (see FIG.
1).
[0053]
FIG. 3 shows the sound pressure generated by the loud sound from each of the speakers So and
Sg1 to Sg4 determined as described above at the sound receiving point Sg4. That is, in this case,
the flesh voice from the speaker O arrives as the first preceding sound, followed by the loud
sound from the main speaker So closest to the speaker O, and then the loud sounds from the
peripheral speakers Sg1, Sg2 and Sg3 Reach in order. Finally, the loud sound from the peripheral
speaker Sg4 arrives. Since the relationship between the natural voice and the loud sound from
each speaker and the mutual relationship between the loud sound from the loudspeaker satisfy
04-05-2019
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the condition that the preceding sound effect works, the sound field is synthesized for the
listener 2 of the receiving point Sg4. The subjective level of sound pressure is considered to be as
shown by the solid line in FIG. That is, the solid line in FIG. 4 indicates that the loud sound from
the speakers in a plurality of distributed arrangements is perceived by the listener 2 as one
sound.
[0054]
The present invention uses the speakers having substantially the same acoustic characteristics,
so that there is no difference in the timbre of the speakers, and by setting the speaker spacing
appropriately, the difference in the transmission characteristics for the listener's ears is
accommodated within an acceptable range. Since the time of sound generation and the input
power of the speaker are controlled such that the loud sound from the main speaker becomes the
preceding sound, the sound image can be localized in the direction toward the sound source O at
any position on the sound receiving surface. Further, since the selection of the main speakers and
the ranking of the peripheral speakers can be performed according to the movement of the
position of the sound source, the sound image can be localized to the moving sound source.
Further, it is also possible to localize the sound image to a questioner or the like appearing at an
unspecified position.
[0055]
Thus, it is possible to achieve the provision of the "sound image localization and sound
amplification method and sound amplification system using distributed speakers, which can
change the direction of sound image localization in accordance with the direction of the sound
source", which is the object of the present invention.
[0056]
Preferably, by suppressing the listening sound pressure level of each sound receiving point Pi to
a predetermined target sound pressure level Lhc, even if the listener 2 is at any sound receiving
point Pi of the sound receiving surface Ph, listening at the predetermined target sound pressure
level It can be so.
In this case, in step 611 of FIG. 6, for example, the listening sound pressure level at the sound
receiving point Pg4 is reached in advance with the speech sound reaching sound pressure level
04-05-2019
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L44 (= 10 log I44) from the peripheral speaker Sg4 at the sound receiving point Pg4. It is
determined as a total (= 10 log ((I 44 + I I k 4 + I o 4)) with lower order peripheral speakers Sg k
(k = 1, 2, 3) and the sum total of loud sound arrival sound pressure levels from the main speaker
So (= 10 log (Σ I k 4 + I o 4)). When the obtained listening sound pressure level exceeds the
target sound pressure level Lhc, the total arrival sound sound pressure level L44 of the
peripheral speaker Pg4 within the range where the preceding sound effect can be obtained (= 10
log ((I44 + Ik4 + Io4)) is the target sound pressure level Suppress until it matches Lhc.
[0057]
More preferably, the target sound pressure level Lhc is determined as shown by the solid line in
FIG. 5, and the sound pressure level of the synthetic sound field on the sound receiving surface
Ph created by each speaker Sj is controlled to attenuate as it gets away from the sound source O.
Do. This attenuation control enables the formation of a sound field consistent with the everyday
experience that the sound decreases as it leaves the sound source O, and works so that the sound
field is perceived as natural to the listener. Therefore, in the case of a lecture or the like, the
listener can listen while having the feeling that the speaker is speaking, and the speaker can feel
psychological feedback such as expanding the topic while feeling the reaction of the audience
well It can be expected that the participants will be able to communicate with each other.
[0058]
The target sound pressure level Lhc shown in FIG. 5 decreases in level according to the distance
Rh from the sound source O, but the inclination of the reduction with respect to the distance Rh
is smaller than the inclination of sound attenuation in the free sound field. According to this
target sound pressure level Lhc, it is possible to perform loud sound with a sufficiently high
sound pressure while giving a natural perception to the listener. The black portion of 40 dB or
less in the figure indicates a range in which the sound is too small to interfere with the listener's
hearing. In a free sound field, even when the speaker utters a loud voice, it may be difficult for a
listener at a distance of about 100 m to hear, but according to the sound amplification method
using the target sound pressure level Lhc in FIG. You can hear the speaker's voice with a
sufficient sound pressure even at a point 100m away while giving a sense of sound field.
[0059]
04-05-2019
22
The inventors set the target sound pressure level Lhc as the target sound pressure level Lhc, and
set the loud sound arrival sound pressure level Loo from the main speaker So at the sound
receiving point Po below the main speaker So as the maximum level. By using an inclined target
sound pressure level Lhc = Loo−3.0 × log Rh where the level decreases in proportion to the
logarithm of the distance Rh (log Rh), the listener does not feel an unnatural gap between the
sound and the distance It has been experimentally confirmed that a sound field can be formed
which can hear clear and clear natural sounds as if the speaker is nearby even at a position
distant from the speaker. However, the proportional constant of the attenuation of the inclined
target sound pressure level Lhc is not limited to the illustrated example.
[0060]
DETAILED DESCRIPTION OF THE PREFERRED EMBODIMENT FIG. 12 shows an embodiment of a
block diagram implementing the present invention using a commercially available monaural
signal conditioning unit 20a. In this embodiment, the signal conditioning device 20a and the
main amplifier 12 need to be provided separately for each speaker Sj. For this reason, an audio
signal distributor 13 for distributing the signal from the signal transmission device 10 to each
speaker Sj is inserted between the mixer 11 and the signal conditioning device 20a. Also, the
computer 30 needs to have control lines for each signal conditioning device 20a.
[0061]
Further, FIG. 13 shows another embodiment of the present invention using a composite amplifier
20b in which the signal conditioning apparatus 20 and the amplifier 24 are integrated, and FIG.
14 shows a block diagram of the composite amplifier 20b used in the same embodiment. In the
illustrated example, the composite amplifiers 20b are connected in tandem, and an audio signal
and a control signal are supplied to each of the composite amplifiers 20b by one digital audio
signal line 27 and a control signal line. The composite amplifier 20b has two acoustic delay units
21L and 21R and two volume control units 22L and 22R, and can simultaneously process digital
audio signals of two channels.
[0062]
The tone generation time (delay time) and input power (volume) of the main speaker So and each
of the peripheral speakers Sgx calculated by the computer 30 (see FIG. 15) are sent to the CPU
04-05-2019
23
25 of the composite amplifier 20b via the interface 14 and control line 28. . The CPU 25 is
connected to the acoustic delay units 21L and 21R and the volume control units 22L and 22R,
and adjusts the sounding time and input power of the digital acoustic signal of two channels
according to the processing pattern stored in the pattern memory 26. The adjusted digital audio
signal is mixed by the mixer 23 and sent to the speaker via the amplifier 24 and the output line
29. By using the composite amplifier 20b of FIG. 10, the sounds of two sound sources can be
processed simultaneously, and sound images directed to two sound sources on the sound
receiving surface Ph can be localized simultaneously.
[0063]
FIG. 15 shows still another embodiment of the present invention using the composite amplifier
20b. Fourteen speakers Sj were dispersedly arranged in a grid at intervals of 5 m × 3.5 m at a
height of 3.5 m from the floor surface, and the composite amplifiers 20 were attached to the
respective speakers Sj. As shown in the figure, the sound of the first sound source below the
speaker S1 and the sound of the second sound source below the speaker S4 are collected by the
microphones 3 and 4, respectively, and input to the interface 14 via the mixer 11. On the other
hand, in the computer 30, control signals necessary for sound image localization to two sound
sources are respectively calculated according to the flow chart of FIG. The digital sound signal
from the mixer 11 and the control signal from the computer 30 were transmitted to the
composite amplifier 20 of each speaker Sj via the interface 14. The composite amplifiers 20 were
connected in tandem as shown in FIG. The composite amplifier 20b simultaneously processes the
digital sound signal from the microphone 3 sent to the right channel and the digital sound signal
from the microphone 4 sent to the left channel to the respective speakers Sj via the mixer 23 and
the main amplifier 24. I output it.
[0064]
Table 2 shows an example of control signals used in the embodiment of FIG. The inventor
confirmed that the control signals in Table 2 can simultaneously localize the sound image
directed to the first sound source and the sound image directed to the second sound source on
the sound receiving surface Ph. As can be seen from the example of FIG. 15 and Table 2, even
when two sound sources O exist in the sound field, such as a speaker on stage and a person on
the floor, for example, the loudspeaker method of the present invention For example, it is
possible to perform sound image localization with the speaker as the first sound source and
simultaneously perform sound image localization with the person as the second sound source. By
performing two sound image localizations at the same time, the speaker and the questioner can
04-05-2019
24
listen to each other's voice well, so it is easy to talk. Furthermore, the sound that the interaction
between them can be heard naturally from anywhere in the venue You can create a place.
[0066]
As described above in detail, according to the sound image localization method and the sound
amplification system using distributed speakers according to the present invention, a speaker Sj
having a plurality of substantially the same acoustic characteristics is covered in an area above a
sound receiving surface. Of the speakers Sj, the speakers closest to the sound source O are the
main speaker So, and the remaining speakers are the peripheral speakers Sgx ranked in
ascending order of distance from the main speaker So. The sound generation time and input level
of each peripheral speaker are controlled so that the loud sound from the main speaker gives the
preceding sound effect at the sound receiving point below the peripheral speakers, and the
localization direction of the sound image on the sound receiving surface is directed to the sound
source O As it is a direction, it produces the following remarkable effects.
[0067]
(A) Among the distributed speakers, the speaker closest to the sound source is selected as the
main speaker, so even when the sound source moves, the sound image is directed in the direction
toward the sound source according to the movement by reselecting the main speaker. Can be
localized.
(B) Since speakers having substantially the same acoustic characteristics are used, differences in
timbres from the speakers do not disturb sound image localization, and sound image localization
in the sound source direction can be obtained anywhere on the sound receiving surface.
[0068]
(C) A plurality of sound images can be simultaneously localized in the sound field by processing
sound signals of different sound sources according to each channel using a signal adjustment
device or the like compatible with multiple channels. For example, the sound image is localized as
the first sound source and the sound image is localized as the second sound source, so that the
exchange between the speaker on the stage and the people in the hall can be performed
smoothly. (D) Since the speaker closest to the sound source is the main speaker and the sound of
the sound source is returned to the vicinity of the sound source, the main speaker can be a so-
04-05-2019
25
called "speaker speaker". Since the speaker can always hear the bouncing sound wherever he is
on the sound receiving surface, he can always speak while checking his own voice, so that he can
feel that he can speak comfortably.
[0069]
(E) By adjusting the listening sound pressure level at each sound receiving point so as to obtain
attenuation corresponding to the distance from the sound source, it is possible to create a natural
sound field according to the listener's daily experience, It can enhance the realism of the sound
field. (F) Since a natural and highly realistic sound field can be created, it is possible to expect an
effect that the emotional expression through the music of the performer can be felt with a sense
of presence at a concert etc. It can be expected to listen while having the feeling that it is being
spoken. Also, the speaker side can naturally receive the reaction of the audience such as the
questioner, and it can be expected to be used to improve the communication between the
speaker and the participants and the participants.
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26
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