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JPH039226

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DESCRIPTION JPH039226
[0001]
The present invention relates to an acoustic apparatus capable of varying physical quantities so
as to enable more favorable music listening, focusing on converting the acoustic characteristics
of a sound field to listen to music or the like into the form of physical quantities and
psychological quantities that can be easily evaluated. It is a thing. Although research has been
conducted on the preferred sound and sound field in the past, the research conducted this time
has set forth the following parameters necessary to determine a good sound field in the concert
hall, which is the basis of the reproduced sound of the audio device. It has gradually become
clear that there is something to show. That is, important objective parameters representing the
characteristics of binaural sound are four elements of listening sound pressure, delay time of first
reflected sound, reverberation time of subsequent reverberation, and interaural cross correlation
coefficient This was revealed by a test of depression (hearing comfort) in a series of simulated
sound fields. Next, the above four elements will be described in detail. First, FIG. 1 shows the
relationship between the sound source and the human head in the space where the reflecting
wall exists. In the figure, αυ is a human head, α power is a sound source, and 030 is a
reflection wall. Here, assuming that the sound source signal is p (t) and the impulse responses
from the sound source to the left ear and the right ear are hp (t) · hr (t), respectively, the signal fz
(t) of the left ear and the right ear f and (D are respectively fz (t) = fo "p (v) h 仝 (t-v) d v = p (t) * l12 (1) (formula 1a) f, (t) = fo ' p (L /) h, (t−L /) dCee I) (t) * h, (t)... (弐 1 b). The symbol * in the
above equation indicates convolution. In the figure, n = 0 directly enters the ear, n = 1 reflects
sound reflected by the reflecting wall αQ enters the ear, n = 2 reflects sound reflected by the
reflecting wall α In the case of entry, m = 1 indicates the case where sound enters the left ear,
and m = r indicates the case where sound enters the right ear. By the way, in FIG. 1, although the
sound which enters the ear after being reflected by the reflecting wall a ′: a C14) is shown only
for one reflecting wall, it is assumed that a large number of such reflections occur. Assuming that
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the impulse response at the time of reflection in the wall α30 is W n (t), the impulse responses
reaching the left ear and the right ear (t), h, (t) are respectively hp (t) = Σ G, W, (-Δt,) * h, 1 (t) n
= 0 ... (Equation 2a) h, (t) = G G, W, (-Δt,) * h, ( t) n = 0 · · · (Expression 2b).
If this (formula 2a) and (formula 2b) are used, above-mentioned (formula 1a) and (formula 1b)
will be fz (t) = sigma p (t) * G, and W, (t-deltat), respectively * H, (t) n = 0 (formula 3a) f r (t) = p p
(t) * G n W, (t-− t) * h, 1 (t) n = 0 · · · Formula (3b) Here, when the sound source p (t) does not
have radiation characteristics such as-, p (t) is p in consideration of radiation patterns different in
each direction. It can be replaced by (r). By the way, an independent and objective acoustic
parameter is included as the information in the acoustic signal entering both ears, and the sound
source signal p (1) can be mentioned as the first parameter. The long-term autocorrelation
function p p (τ) can be expressed as (Equation 4) using the sound source signal. Put)-p (t) * s (t)
where s (t) corresponds to the sensitivity of the ear and is theoretically represented by the
characteristics of the middle ear and the outer ear Can be expressed as an impulse response of a
G filter well known as an approximation of. Naturally, the normalized autocorrelation function φ
p (τ) can be expressed as p p (o) by dividing (Equation 4) by the power p p (0) of p = (t). FIGS. 2
(a) and 2 (b) show the measured values of the normalized autocorrelation function corresponding
to the above (Equation 5). FIG. 2 (a) shows measured values corresponding to the music "Royal
Pavane" by Gibbons, and this music is hereinafter referred to as music A. Fig. 2 (b) shows the
measurement values corresponding to the music "Shinhonietta 9: 48: Movement, Allegro con brio
(Synfornietta, 0pus 48: m Movement, Allegro con brio)" by Arnold. Music is hereinafter referred
to as music B. Next, a second objective parameter is an impulse response generated by reflection
at a boundary such as a wall. This relates to the initial time delay between the direct sound and
the first reflection sound, and also relates to the change of the spectrum based on the initial
reflection sound, the subsequent reverberation sound, and the reflection. The third objective
parameter is impulse response to the left and right ears, z (t), h, (t). The impulse response plays
an important role in sound localization and is not independent of each other.
That is h, z (t), y h in the case of a centrally localized signal. It becomes clear from becoming (t).
Next, take these two impulse responses h, v (t). , (T) derive the mutual dependency between them.
First, a long binaural cross correlation function 09. 9 between the binaural signals fz (t), f, (t).
Expressing (τ) using f or (t), f, (t), τ1 ≦ Ims (Equation 6) By the way, the sense of spread or
direction is a value with a small interaural cross correlation The binaural cross-correlation
function has a large peak at 1τ1 <1 ms only in the case of a signal from a fixed direction. The
time difference between the signals fv (t) and fr (t) between the two ears can usually be 1 ms or
more from the relationship between the interaural distance and the speed of sound. It is because
it is not. First, the binaural cross-correlation function 又 or (τ) by direct sound only fx (t) = p (t) *
h, t (t). It can be obtained by substituting fr (t) = p (t) * h, (t) into (Expression 6). By the way, the
normalized autocorrelation function φ intersection of only direct sound, (τ) can be expressed as
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(Expression 7). In addition, if φ statement, (τ) ° to him h, v (t) = h, (t), it becomes almost l.
Here, ¥¥ ′ ′ x (o) and Φ, '; (o) indicate the τ-0 autocorrelation function of the signal in each of
the left and right ears. Next, if it is assumed that another reflected sound is added to the direct
sound after the time when the direct wave autocorrelation function becomes smaller, then the
normalized autocorrelation function φ2. In the case where W and (t) are equal to the delta
function δ (1) of Dirac, (τ) is expressed as (Equation 8). Here, Φ or (τ) is the inter-ear crosscorrelation function of the n-th reflection, Φ n ′ ′ ff 1 (o), Φ r: ′ (0) is the τ of the n-th
reflection of the left and right ears The autocorrelation function at = 0 is shown. When the sound
source is in front of a normal room, the maximum value of the interaural cross-correlation
function is obtained extremely near τ = 0. By the way, the strength of the normalized interaural
cross correlation is defined as IAAC as IAAC = 1.phi.g, (r.sub.1) max forl r.sub.l.ltoreq.lm5
(Equation 9). In addition, φ2. (Τ) = Φ cross, (τ) 7 ° C. 11 ′ ′ Y is the correction, 目 eyes (0),
Φ 2, (0) indicate the autocorrelation function at τ = 0 of the signal sound of the left and right
ears respectively ing. In the above, a long-term autocorrelation function of the sound source
signal, a plurality of impulse responses generated by reflection of sound by a wall or the like, and
an interaural cross correlation function showing a correlation between binaural signals are
described. From these physical quantities, four factors are important objective parameters that
characterize the sound of binaural signals: listening sound pressure, delay time of the first
reflected sound, reverberation time of the subsequent reverberation We will describe how to find
the optimal value of each of the interaural correlation function values.
The results of the bias test performed using the music A and music B described above are shown
in FIGS. 3 (a) and 3 (b). The horizontal axis is IACC1 and the vertical axis is listening sound
pressure (unit: dBA). The listening sound pressure can be considered to be the value of τ = 0 of
the autocorrelation function, and the value on the horizontal axis of the graph indicates a bias.
Fig. 3 (a) (The listening sound pressure [p) p which is clearly more optimal than bl is not so
dependent on IACC, and in the case of music A with a somewhat slow tempo, 77-79 dBA; It turns
out that it is 79-80 dBA in the case. In any case, it can be seen that the optimum value of the
listening sound pressure is around 79 dBA. Next, according to the result of evaluating the
synthetic sound field consisting of the direct sound and the single reflection sound at the time of
speaker reproduction using music and speech, it is found that the normalized autocorrelation
function φp (τ) of the sound source signal When the level of the reflected sound is changed
over ± 6 dB of the direct sound, the optimum delay time of the reflected sound is that 1pp (τ)
corresponds to 1/10 of the level G1 of the first reflected sound. It became clear that it
corresponded to time. Therefore, time τ d corresponding to 1/10 of the level G 1 of the first
reflected sound is taken on the horizontal axis with 遅 れ p (τ) 1 being 1/10 of the level G 1 of
the first reflected sound, FIG. 4 is an axis view. The range shown in the figure shows the delay
time at 0 and 1 lower than the maximum value of the bias, and the symbol O in the figure is the
level G of the first reflected sound, G = 6 dB,. OdB, the mouth has shown each at GI = -6dB. In
particular, if it is assumed that the time when 1φp (τ) 1 becomes 0.1 times φp (0) is τp, then
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it can be expressed as τd = τp when Gl = O dB. Incidentally, the τp of the musics A and B
described above is 127 m5 and 35 m5, respectively, as can be deduced from FIGS. 2 (a) and 2
(b). Here, as is apparent from FIG. 4, a straight line can be drawn by approximating each point in
the figure, and when looking at both axes corresponding to the straight line, τ d is a single for
which the block is the largest. It can be seen that the delay time [Δt +) p of the reflected sound
substantially matches. At the same time, the time τρ at which 1Φp (τ) 1 becomes 0.1 times
Φp (0) is also approximately the same. That is, if this is expressed by a mathematical expression,
the delay time of the first reflected sound [.DELTA.t1] p is the most preferable [.DELTA.t,] p =
.tau.p (10% formula% .PHI. (. Tau.) 1.ltoreq.KG '... (Expression 11) When τ> τp, it can be
expressed as 1φ p (τ) 1 ≦ 0.1 (Expression 12).
Furthermore, the autocorrelation function of the sound source signal is closely related to the
optimum reverberation time, and this relation is shown in FIG. The vertical axis is the optimum
reverberation time (Tsub) p of the subsequent reverberation sound, and the horizontal axis is
τp. The reverberation time referred to here is not the time when the direct sound attenuates by
60 dB, but it is expressed as the time when the signal of the reverberation part attenuates by 60
dB. In the figure, music A and music B are the same as those described above, but music E is
"Symphony in C major, 551 Shijubita 1st fourth movement, Malto allegro" by Mozart (Mostar),
and the speech S is It is surprising that the sky is blue, bright and high, regardless of the sound of
"the sound of Tonese for Tone" by Doppo Kunikida. It is J ((tau) p = 12 ms). As is apparent from
the figure, the relationship shown in FIG. 5 can be approximately approximated by the function
(Tsub1 p + (23 ± 10) rp). Next, FIG. 6 shows the results of measurement in a synthetic sound
field consisting of direct sound and single reflection sound, where the horizontal axis is the
arrival direction ξ of the reflected sound and the vertical axis is the vertical axis. Shows the
value of the reference and IACC. It can be seen from FIG. 6 that the reference increases as the
value of IACC decreases. That is, the correlation function between the score of the reference and
the value of IACC is negative (-0, 76: 1% significant level), which holds when IACC takes a
maximum value at τ = 0 It is a thing. It can also be read from the figure that in order to reduce
ICAA most effectively, the initial reflection sound should be made to come within the range of ±
(55 ° ± 20 °) from the front. Next, the measure of the bias by the four parameters described
above will be described. By the way, the measure of this reference is determined by a
comparative test, and each parameter affects the measure of preference independently. As a
result, since the theory of superposition can be applied, it is possible to generalize the data of the
sound field block obtained in the concert hall or the like by normalizing each objective parameter
with the optimum value. The following describes the measures of the block obtained by the
comparison test. First, the measure S1 of the bias as a function of the listening sound pressure is
shown in FIG. In this figure, the scale at the optimum listening sound pressure is set to zero. The
value of Sl is a left and right target centering on the most preferable listening sound pressure
(OdB), but it is slightly weaker when it is shifted to the listening sound pressure than when it is
shifted to the strong sound pressure. There is a tendency for the measure of the bias to be good.
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If this is expressed by a formula, 5t (LL) S S + (LL) (formula 13)% formula%)), p is the listening
sound pressure, and (p) p is the optimum listening sound pressure. In the figure, ○ indicates
music A, and X indicates the value of music B. Next, a measure S2 of the bias as a function of the
delay time (first reflected sound delay time) between the direct sound and the first reflected
sound is shown in FIG. The horizontal axis of this figure is normalized by the most preferable
time delay (Δt +) p. By the way, it is known that the delay time of the second reflected sound is
that the optimum value [Δt,] p is [Δtz] p zl, 8 rp. This optimal value was used as the delay time.
Of course, it is needless to say that there is the condition of (Expression 10). In the figure, each of
○, a and A △ is an individual measurement result by music A, each of X, b and B △ is an
individual measurement result by music B, C is music C and D is music D , The mouth is music E,
and the measurement result by speech S. Music A, B and E have already been mentioned above,
but Music C is Hyton's Symphony 102 No. 2nd movement Adagio (τp = 65 ms), and the music is
Wagner's Seek free toy dill (SiegfriedIdyll) It is 332 measures (rp = 40 ms). As in FIG. 6, the scale
in the delay time of the optimal first reflected sound is set to zero. Furthermore, it is a preference
preference measure S as a function of the reverberation time of the subsequent reverberation,
and the dashed line is that for G-1, l. In the figure, ○ and a are for music A, X and b are for music
B, the mouth is for music E, and · are for experimental results for speech S, and both are for G =
4.1. . G = 4.1 corresponds to the case where there is much reverberation as in the rear of the
concert hall, and G = 1.1 corresponds to the case where there are many direct sounds as in the
front seat of the hall. The preference scale at the most preferred reverberation time is set to zero.
Next, the measure S of the bias as a function of IAcc is shown in FIG. In the figure, ○ is an
experiment result for music, X and b are results for music B. Due to the nature of IACC, the sound
image is in the frontal direction if its maximum value is taken at τ = 0. As is apparent from the
figure, as l5CC approaches 1, the measure S4 of the bias rapidly decreases. Therefore, IACC
should be smaller than 0, 5 as much as possible.
So far four measures S of discount! To S4, these four measures can be approximated by the
following approximate equations. First, the scale St of the block of the listening sound pressure
shown in FIG. The scale S2 of the equation% Bref can be expressed as S2 = −α21 × 21 ′ ′ ′
′ where X2 = log (Δt1 / [Δt +] p). (Equation 17) ... (Equation 18). Further, the measure S3 of
the preference of the reverberation time of the subsequent reverberation shown in FIG. 9 is 53-.alpha.31xil "'(Eq. 20) where X3 = log [Tsub / l: Tsub) p)-((20) Formula 21)% Formula% Next, the
delay time of the first reflected sound shown in FIG. 8 is ... (Formula 22) If α3 becomes negative
in (Formula 22), α3 = 0 ... It is set as (Formula 22 '). The rest of the scale S4 of the biases of IACC
shown in FIG. 10 is 54 =-. Alpha.4.times.4 "/ 2 (Equation 23) where X <= IACC (Equation 24)%
equation% .alpha.4 = 1. 45 ± 0.44 (Expression 25). Based on the principle of superposition, S = 5
5i = 1 (Equation 26)% Formula% (12, 3, 4) as the total scale S of the preference in the concert
hall etc. The scale is shown. An accurate evaluation of the sound field can be performed by using
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the measures S and S + (i = L 2, 3.4) of the references thus obtained. This invention uses four
parameters: listening sound pressure in the sound field, delay time of the first reflected sound,
reverberation time of the subsequent reverberation, and interaural cross-correlation coefficient
to measure the blockage in the concert hall or room Focuses on the ability to determine the scale
of each of the references and the overall brightness scale, and provides a sound device capable of
correcting the sound field based on the respective parameters to create a more preferable sound
field. Is the purpose. (1) Sound Field Evaluation Meter Related to the Present Invention First, the
sound field evaluation meter related to the present invention will be described using an
embodiment. 11 and 12 are block diagrams showing the general configuration of the sound field
evaluation and measurement instrument. FIG. 12 is a block diagram showing every four
parameters.
In the figure, (1) is a human head or a dummy head, (21) (2r) is an outer ear, and a microphone
inserted in the entrance, (3) (3 j? (3r) is a preamplifier, (4) is a physical quantity analyzer, (5) is a
comparator, (6) is a mental quantity converter, (7) is an overall evaluator, and (8) is an output
terminal of the block , (9) is a recorder for recording a reference, Oo) is a sound field evaluation
instrument, an example is an adder, R is a listening sound pressure analyzer, and (conversion) is
a first reflected sound delay time analyzer , (C) is a subsequent reverberation reverberation time
analyzer, (C) is an interaural cross correlation function analyzer, GD is a listening sound pressure
psychology converter, ■ is a first reflection sound delay time psychology converter, ω Is a
subsequent reverberation reverberation time psychology converter, the figure is a interaural
cross correlation coefficient psychology converter, (y) (y +) (Y2) (Y3) (Y4) is a comparison data
input terminal. The respective optimum values are inputted to the comparison data input
terminals (y) (y +) (yz) (yz) (y4), and the optimum listening sound pressure ((p) p) is inputted
from the input terminal (yl) The corresponding signal corresponds to the optimal first reflected
sound delay time ([Δt +) l] from the input terminal (y2), and the last subsequent reverberation
reverberation time (CTsub) p from the input terminal (y3) From the input terminal (y4) is a signal
corresponding to the optimal interaural cross correlation coefficient. Note that the listening
sound pressure analyzer CD in Fig. 12 · The first reflection sound delay time analyzer · The
subsequent reverberation reverberation time analyzer (C) · the interaural cross correlation
function analyzer 卿 is the physical quantity analysis in Fig. 11 Corresponds to the physical
quantity analysis unit corresponding to the device (4). Listening sound pressure psychology
converter G υ · first reflection sound delay time psychology converter in Fig. 12 · · · · ·
subsequent reverberation reverberation time psychology converter 弥 · interaural cross
correlation coefficient psychology converter (support) Corresponds to a comparison and
psychology converter corresponding to the comparator (5) and the psychology converter (6) of
FIG. Similarly, the comprehensive evaluator (7) corresponds to the comprehensive evaluation
unit. First, a simple flow of one embodiment of this sound field evaluation measuring instrument
will be described with reference to FIG. First, install a dummy head or head (1) at a place where
you want to evaluate the sound field, such as a concert hall or a listening room. The sound
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pressure signals of both ears are detected by the microphones (2β) (2r) attached to the left and
right ear canal entrances, amplified and added by the pre-amplifier (3), and then the physical
quantity analyzer (4 Physical pressure, ie, the listening sound pressure, the first reflected sound
delay time, the subsequent reverberation reverberation time, and the interaural cross correlation
coefficient IACC, from the sound pressure signals of both ears according to the respective values
in the comparator (5) Psychological amount conversion to be performed based on the respective
optimum values input as comparison data from the input terminal (y), and then the processing
based on (Equation 14) (Equation 17) (Equation 20) (Equation 23) The amount is converted into
a mental amount, that is, an amount corresponding to a reference by the unit (6), and then
comprehensively evaluated in the total evaluation unit (7) to output the reference in the sound
field from the output terminal (8).
This calculator can be used to evaluate the sound field of a concert hall or a listening room. Next,
the embodiment shown in FIG. 11 will be described in detail with reference to FIG. 12 which
further shows the flow of four individual parameters. As described above, the sound pressure
signal detected by the microphones (1) and (2r) is amplified by the preamplifier (3f) and the
adder! After being added by 4 ■, it is input to the listening sound pressure analyzer CD
corresponding to the physical quantity analysis unit, the first first sound emission period
analyzer, and the subsequent reverberation reverberation time analyzer さ れ, and their
respective analysis Listening sound pressure p, the first shot sound delay time j in the device!
Each of the subsequent reverberation reverberation time Tsub is analyzed and measured. Also,
apart from these three flows, the outputs from the two preamplifiers (31) (3r) before being added
by the adder Q @ correspond to the interaural correlation function analysis corresponding to the
physical quantity analysis unit The input value is input to the device (R), and the maximum value
IACC of the interaural cross correlation coefficient is analyzed and output. Next, the four sound
analyzers corresponding to the physical quantity analysis unit (listening sound pressure p
outputted from each of 41 (conversion)), the first shot sound delay time Δjl + the subsequent
reverberation reverberation time Tsub, the maximum Interaural cross correlation coefficient
IACC is the listening sound pressure psychology converter 61) corresponding to the comparison
and psychology conversion unit, the first reflection sound delay time psychology converter
mouth, the following remaining one sound reverberation time psychology Optimal listening
sound pressure [: p) p, optimal first reflected sound delay time [Δ] input to the converter group
and input from each of the other comparison data input terminals (y +) (yz) (y3) (y4) t + 31), the
optimal subsequent reverberation reverberation time [Tsub) p, and the optimal interaural cross
correlation coefficient are compared. Although the optimal interaural cross correlation coefficient
may be considered to be 0, there is often no problem in setting it to 0.4 to 0.5 or less in practice,
so in this embodiment it is set to 0.4. ing. In the comparison and the calculation formula for the
psychology conversion in the psychology conversion part, in the listening sound pressure
psychology conversion part (51), the aforementioned (Equation 14) to (Equation I6), the first
reflected sound delay time psychology quantity In the converter ω, the above-mentioned
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(Equation 17) to (Equation 19), and in the subsequent reverberation reverberation time
psychology converter I, the above (Equation 20) to (Equation 22-), interaural cross correlation
coefficient psychology In the quantity converter (2), as shown in (Equation 23) to (Equation 25),
the measures S1, S2, S3 and S4 of the block which are the outputs obtained from this comparison
and the psychology conversion unit It can be determined by calculation processing by a
microcomputer program or the like.
However, if the accuracy is not a problem, a conversion table as shown below may be used.
Conversion table ■ LL (dB) 10, 7 10, 5 10, 3 10, 20, IO 20-0, 1-0, 2 10, 3 10, 5 10, 8-1. 2LL =
2010 g (p / [1]:] p) conversion table {circle over (1)} 1 + / Δt +) p
20.10.20.30.40.50.60.70.80.92.03.04 05. 06.07.08.09.00.00-1.10, 6, 10, 4, 20, 1-, 0-0.0-0.0 10,
00, 0 1 、, 2 4−, 4-0.7-0.81, 0-1.2-1.3-1.4-1.5 The above conversion tables 1 and 2 are listening
pressure and psychological quantity converter 1! It is an example of a conversion table of in and
a 1st reflected sound delay time psychology converter figure. In the conversion tables I and II,
the listening sound pressure p, the optimum listening sound pressure (pop, the scale S + of each
of the references corresponding to the values of the first reflected sound delay time Δt1 and the
optimal first reflected sound delay time, St is stored. Although only the conversion table of the
listening sound pressure psychology converter (51) and the first reflected sound delay time
psychology converter ω is shown here, the subsequent reverberation reverberation time
psychology converter ■ and the interaural cross correlation A similar conversion table can be
created and carried out with respect to the number psychological quantity converter diagram. In
this way, the comparison and psychology converter converters (51) to the bias measures 81 to
S4 output from the control are input to the comprehensive evaluator (7) and obtained as a result
of the comprehensive evaluation. A measure S of the overall bias is output at the output terminal
(8). This output is recorded on the recorder (9). However, in this embodiment, although the
recorder (9) is included, it is not necessary to necessarily include the recorder (9), and it is
needless to say that it may be changed to a display. Next, the listening sound pressure analyzers
α0... Of the physical quantity analysis unit used in the sound field evaluation measurement
instrument 00) of FIG. For the first reflected sound delay time analyzer, Fig. 13 (a) to (e) for an
example of the respective configurations of the subsequent reverberation reverberation time
analyzer (2) and the interaural cross correlation function analyzer (good) It demonstrates using.
Fig. 13 (al is the first reflected sound delay time analyzer using a square integral type
reverberator with M, R, Schroeder (M, R, 5 chroeder modified al) or the subsequent reverberation
reverberation time analysis It is a figure showing the basic composition of equipment, and it has
a function of a listening sound pressure analyzer reconnaissance that an output corresponding to
sound pressure is also included.
FIG. 13 tb + shows an example of a priority encoder used to output a code corresponding to the
first reflected sound or reverberation time of the first reflected sound delay time analyzer □□□
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or the subsequent reverberation reverberation time analyzer FIG. FIGS. 13 (C) and (d) are
diagrams showing other examples of the listening sound pressure analyzer. FIG. 13 (e) is a
diagram showing a configuration example of a circuit for obtaining the interaural cross
correlation coefficient. In FIG. 13 (a), (400) is an input terminal of a circuit used as a first
reflected sound delay time analyzer using a square integral type reverberator or a subsequent
reverberation reverberation time analyzer, (401) (402) ) (403) is a delay circuit, (404) to (411) is
a multiplier, (412) to (4) 5 is an integrator, (416) to (418) is an adder, (419) to (421) Is the
comparator, (422) the autocorrelation coefficient φ. Output terminals (423) to (425) output
information on the delay time of the first reflected sound or the subsequent reverberation
reverberation time, respectively. (426) is the attenuator in FIG. 13 Tb), (430) is the priority
encoder to which 01 to o + i-1 outputted from the output terminals (423) to (425) are input, X1
to X Xn- + is a code output corresponding to the first reflection sound delay time or the
subsequent reverberation sound time, in FIG. 13 (C) (d), (435) is a rectifier circuit, (436) is a
reduction pass filter, (437) Is the multiplier, and in FIG. 13 (e), (441) is the input terminal for the
left input signal Lin of the circuit for determining the interaural cross-correlation coefficient,
(442) is the same (input for the right input signal Rin). Terminals (443) and (444) are adders,
(445) to (448) are root mean square circuits (RMS), (449) is an arithmetic circuit, and (450) is an
output for taking out an output corresponding to the cross correlation coefficient Terminal A. In
the circuit of FIG. 13 (a), (n-1) delay circuits, adders and comparators respectively, and output
signals from the multiplier, integrator and integrator to which the output signals from the delay
circuit are input Although there are n multipliers and n output terminals to be input, in the
drawing, (n-1) are representatively represented by three and n by only four. The remaining ones
are omitted, and the ones drawn in the figure have continuous symbols. Therefore, as the (n-1) th
delay circuit (403) following the second delay circuit (402), the third to the (n-2) th delay circuits
are not illustrated for convenience. First, FIG. 13 [a] will be described.
When a signal is applied to the input terminal (400), the input signal is applied to the multipliers
(404) to (407) and also to the delay circuit (401). The output of the delay circuit (401) is given to
the multiplier (405) and also given to the next delay circuit (402) to produce the delayed signal
up to the output of the (n-1) th delay circuit (403) . The outputs of the respective delay circuits
(401)-(402) are multiplied by multipliers (405)-(407). The multiplier (404) performs a square
operation on the input signal itself. Next, the output signals of the multipliers (404) to (407) are
respectively integrated in the integrators (412) to (415), and the output signals are further
squared by the multipliers (408) to (411). The output signal of the integrator (412) is output as it
is to the terminal (422) separately from the input to the multiplier (408). This signal is the
autocorrelation coefficient φ (0), that is, the square of the power of the input signal, and includes
information corresponding to the sound pressure signal. The processing of the square of power
can have an equivalent function simply as absolute value conversion. By the way, the output
signals of the multipliers (40g) to (411), that is, the square of the autocorrelation coefficient φ.
2φ12. ... The signal of φ 2 o −1 is applied to the adders (416) to (418), and the outputs of the
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adders (416) to (417) are respectively applied to the next adder, (n − 1) to the second adder
(418). By the way, the output signal from the multiplier (408) and the output signals from the
adders (416) to (417) are respectively added to the last adder (418) in the comparison circuits
(419) to (421). Are compared with the signal after being attenuated by the attenuator (426), and
the respective output signals 01 to (423) to On- + (425) are output to the output terminals (423)
to (425). The attenuation ratio of the attenuator (426) is set to (0, 1) 'to obtain the first reflected
sound delay time, and to (0, 01 1)' to obtain the subsequent reverberation reverberation time Is
used. This is similar to what is already known as the optimum delay time detector, and is for
finding the delay time at which the value of the autocorrelation function becomes 1710 or
1/1000. The total delay time from the delay circuit (401) to the delay circuit (403) is required to
be about the length of the first reflected sound delay time and the subsequent reverberation
reverberation time, that is, about 100 to 200 m5 and about 3 to 5 SeC.
Of course, the integration circuits (412) to (415) need to be reset with the passage of time.
However, the reset circuit is not shown. Next, the priority encoder (430) shown in FIG. 13 (b) is
the output signal of the comparators (419) to (421) shown in FIG. 13 (al), (423) to .smallcircle.,-, (
425), and the delay time τ (−τp) corresponding to the attenuation ratio (0,1) ′ or (0,011) 2
of the attenuator (426) of FIG. The code X1... Xn corresponding to the delay time reduced below
the ratio is displayed or output. The following Fig. 13 (C) and (d) are diagrams showing another
circuit example used as a listening sound pressure analyzer, but also in this case, the output from
the output terminal (422) of Fig. 13 (a) Signal φ. Just as you can, you can know the listening
sound pressure. This configuration is a circuit in which a low pass filter (436) and a rectifier
circuit (435) or a square multiplier (437) are combined as shown in FIG. 13 (C). The circuit (519)
for obtaining the interaural cross correlation coefficient shown in FIG. 13 (e) next uses one
example of a circuit method for obtaining a normalized correlation coefficient. First, when the
input signal Lin is input from the input terminal (411) and the input signal Rin is input from the
other input terminal (442), the input signals Lin and Rin are directly squared and averaged (445).
At (448), the respective root mean square values γ and δ are output. Further, the sum signal
between the input signals Lin and Rin is taken out by the adder (443), the difference signal is
made negative by one signal and the adder (444) added to the other signal, and the mean square
circuit (446 (447), and the sum signal and the difference signal are similarly output as the
respective root mean square values α and β. The signals α, β, γ and δ of the four root mean
values obtained in this way are subjected to arithmetic processing in the arithmetic circuit (449),
that is, (α2β2) / (4γδ), and the mutual phase relationship at the output terminal (450) An
output .phi. Corresponding to the number (.degree., That is, the interaural correlation number
(IACC) is obtained. However, when using the circuit of FIG. 13 (e), a delay circuit of 0 to 1 ms is
provided before the input signals Lin and Rin are input to the input terminals 441 and 442, and
the delay time is changed. When the maximum value is determined, the circuit configuration
corresponding to the above (Equation 9) is obtained, and the value faithful to (Equation 9) can be
obtained from the circuit.
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A delay circuit therefor is not shown in FIG. 13 (el. In addition, in order to calculate the interaural
interrelationship number (IACC) with higher accuracy, it is preferable to execute numerical
calculation using the equations (7) to (9) described above. In one embodiment which has been
described so far, the input to the listening sound pressure analyzer CD, the first first sound
sounding interval analyzer and to the subsequent reverberation reverberation time analyzer is a
dummy head or human head ( The sum of the two acoustic signals coming into the left and right
ears of 1) has been used, but the delay time between the two signals is usually less than 1 ms,
and analysis of these three analyzers Aυ Since there is no significant difference in the results,
sufficient analysis results can be obtained using either left or right acoustic signal. The optimum
listening sound pressure [p) p of the sound field evaluation measuring instrument 00) which is
one embodiment of the first invention described so far depends on the type of the sound source,
but they are approximately (79 ± 5) As it is about dB, 79 dB is used here as a representative.
Also, the optimal first reflected sound delay time [Δt + 3 +] and the optimal subsequent
reverberation reverberation time [Tsub] p are first reduced from the actual sound source itself by
the first reflected sound delay time analyzer to less than the attenuation ratio. The delay time τp
may be obtained and set as the optimum first reflected sound delay time [Δt1] p, and that (23 ±
10) times may be used as the optimum subsequent reverberation reverberation time (Tsub) p. In
this embodiment, (Tsub) p = 23 rp is used. As described above, according to this sound field
evaluation measuring instrument, one or two microphones, an amplifier for amplifying an
acoustic signal obtained from this microphone, and a listening sound pressure p from this
amplification signal Listening sound pressure measurement means for measuring the same, first
reflected sound delay time measurement time for measuring the first reflected sound delay time
Δt1 from the increase 1) signal, and also similarly the subsequent reverberation from the
amplified signal The following reverberation reverberation time measuring means for measuring
the sound reverberation time Tsub, the interaural cross correlation coefficient measuring means
for measuring the interaural cross correlation coefficient IACC from the amplification signal, and
the above listening When the listening sound pressure is greater than the optimum listening
sound pressure, the value is 372 power of the absolute value of the value 20 times the base 10
logarithm of the ratio of the sound pressure p to the optimum listening sound pressure [p] p.
(0,07 ± 0.03), when it is smaller And outputs a value that is multiplied by the 04 ± 0.02) as a
measure S1 of blanking reference (psychological goodness of auditory). That is, the first
comparison and conversion means for producing the output corresponding to the above (formula
14) to (formula 16), the above first reflection sound delay time Δt1 and the optimum first
reflection sound delay time [Δt,) p When the first reflected sound delay time is longer than the
optimum first reflected sound delay time to the value of the third power of the logarithm with a
base of 10 as the base,-(1, 42 ± 0, 6) Is smaller, and the value obtained by multiplying-(1, 11 ±
0.5) is output as the measure S2 of the prefense, that is, the output corresponding to the above
(Equation 17) to (Equation 19) The following reverberation reverberation is applied to the value
372 of the absolute value of the base 10 logarithm of the ratio of the comparison conversion
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means 2 and the ratio of the subsequent reverberation reverberation time Tsub to the optimal
subsequent reverberation reverberation time (Tsub) p. When the time is larger than the optimal
subsequent reverberation reverberation time, the sound pressure of the entire reflected sound is
− (0, 0 The sum of the value of (74 ± 0.25) times the value of-(0,45 ± 0.15), and when it is
smaller, the value of-(0,42 ± 0.14) times the sound pressure of the entire reflected sound When
the value after multiplication with (2, 36 ± 0.79) becomes negative, the value is output as it is,
and when it becomes positive, zero is output as the measure S of the bias, that is, the abovementioned (Equation 20) And third comparison / conversion means for producing an output
corresponding to (Expression 22 ') and a value obtained by multiplying the value of the 372
power of the interaural cross-correlation coefficient by-(1, 45 ± 0.44) Fourth comparison /
conversion means for outputting as the reference scale S4, that is, for outputting outputs
corresponding to the above (Equation 23) to (Equation 25), and preferences outputted from the
first to fourth comparison / conversion means Scale S1.
Since S 2, S 2 and S 4 are comprehensively evaluated to provide an overall evaluation means for
outputting a scale S of the whole of the reference, a faithful sound field evaluation can be
performed. (2) The acoustic device of the present invention, based on the following four physical
parameters: listening sound pressure in the sound field, delay time of the first reflected sound,
reverberation time of the subsequent reverberation, and interaural cross correlation coefficient
An embodiment of the acoustic device of the present invention capable of correcting a sound
field and creating a more preferable sound field will be described with reference to the drawings.
(2-1) Acoustic Device of the Present Invention FIG. 14 (a) is a configuration block diagram of one
embodiment of the acoustic device of the present invention capable of correcting a sound field
and creating a more preferable sound field, 14 (bl shows one example of the sound field
expanding apparatus included in the sound apparatus of FIG. 14 (al, and FIG. 14 (C) shows
another example of the reverberation apparatus included in the sound apparatus of FIG. 14 (a) It
is First, in FIG. 14 (a), (501) indicates the input terminal INR1 (502), the input terminal INL,
(503a) to (503d) indicates the adder, (504a) and (504b) the reverberator, (505a) and (505b).
(505r) (505f) is an attenuator, (506a) (506b) is a delay circuit, (507) is a scaler which is a kind of
dividing circuit, (508a) (508b) is an up / down counter, (509) is Digital to analog converter,
(510a) (510c) first reflected sound delay time analyzer, (510b) subsequent reverberation
reverberation time analyzer, (511a) to (511c) encoder, (512a) (512b) (513a) and (513b) are
comparators, (514a), (514b) and (514c) are smoothing circuits, and (515a) and (515b) are for
setting target values Because of the variable resistor, (516) the sound field expansion device,
(517r) (5171) is a power amplifier for driving a speaker, (518R) (5184? ) Is a speaker, (519) is
an interaural cross correlation coefficient (IACC) calculator, (520) is a multiplier, (531) and (532)
is a reverberation device (504a) and (504b), and a sound field enlargement device (516) ) To the
input terminal inr, inl (521) is the CLOCK signal input terminal, (533) (534) is the output from
the sound field expander (516) outr, outf, (535) is the sound from the comparator (513b) It is an
input terminal attm to the field expansion device (516). In FIG. 14 (b), (530a) to (530d) are
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adders, (536) is an attenuator, (537) is a low pass filter (53, 8) is a high pass filter (539) is a
phase shift. It is a circuit or a delay circuit.
FIG. 14 (C1: (504c) M, R, Schroeder (Schroeder) and B, S. The reverberators 54a to 540i by Atal
are adders, 541a and 451b are phase inverting circuits, 542a and 542 are voltage followers, and
543a to 543f are delay circuits, (544a) to (544f) are attenuators, (545) is an input terminal of
the attenuator (504c), and (546) is an output terminal of the attenuator (504C). In the drawings,
the same reference numerals as in the other drawings indicate the same or corresponding parts.
Next, the operation of one embodiment of the acoustic apparatus of the present invention will be
described with reference to FIG. First, record, tape and microphone to input terminals JNR (501)
and INL (502). An acoustic signal from a radio or the like is given. This acoustic signal is added
by the adder (503c) and input to the first reflected sound delay time analyzer (510c). This first
reflected sound delay time analyzer (510c) is constituted by the same circuit as the circuit shown
in FIG. 12, that is, the circuit shown in FIG. 13 (al shown in FIG. 12). The attenuation ratio is set
to 1 / lO so that the reflected sound delay time can be measured. The output of the first reflected
sound delay time analyzer (511) is input to the encoder (511), and the delay time [Δt +] p of the
first reflected sound optimal for the sound source is obtained as τp from the encoder (511). The
output ([Δt +) p = τp) comes out. The output from the encoder (511c) is input to the
comparator (512a) and is also input to the multiplier (520) and multiplied by 23 to obtain the
optimum subsequent reverberation reverberation time (Tsub) p (= 23 ° τp) Are provided to the
comparator (512b). On the other hand, the dummy head placed in a sound field such as a
listening room or the microphones (2r) provided at the left and right ears of the human head (1)
After amplification by (3r) and addition of the output by the adder (503d), the same is applied to
the same first reflected sound delay time analyzer (510a) as in FIG. 12 and It is also input to the
subsequent reverberation reverberation time analyzer (510b). The configuration of the
subsequent reverberation reverberation time analyzer (510b) is the same as the circuit
configuration described in the throat of FIG. Next, the output of the first reflected sound delay
time analyzer (510a) is input to the encoder (511a), and the time corresponding to the delay time
Δt1 of the first reflected sound after the influence of the sound field is obtained. .
Thereafter, it is input to the comparator (512a). In addition, the output of the subsequent
reverberation reverberation time analyzer (510b) is input to the encoder (511b), and a signal
corresponding to the subsequent reverberation reverberation time Tsub after being given the
influence of the sound field is obtained, and this signal is It is input to the comparator (512b). By
the way, in the comparator (512a), the optimum first reflected sound delay time [Δt +) I) from
the encoder (51IC) and the first reflected sound after being given the influence of the sound field
from the encoder (511a) Of the first reflected sound delay time [Delta] t, and if the first reflected
sound delay time [Delta] t1 is greater than the optimum first reflected sound delay time [[Delta] t,
p], the first reflected sound delay time is compared. In order to reduce Δt, the comparator
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(512a) applies a countdown signal CD to the counter (508a) to reduce the contents of the
counter (508a), so that based on the contents of the counter (508a) In the scaler (507) which
performs the cycle, the value to be divided becomes smaller, and the frequency of the DELAY
CLOCK signal obtained by dividing the CLOCK signal from the input terminal (521) becomes
high. , That the delay time of the delay circuit (506a) (506b) configured by the EIBD (packet,
brigade device) etc. in the acoustic circuit (504a) (504b) is reduced by the DELAY CLOCK signal
It is. However, if the first reflected sound delay time Δt1 is smaller than the optimal first
reflected sound delay time [Δt +] p, the signal CU of the count-up from the comparator (512a) to
the counter (508a) , The content of the counter (508a) becomes thicker (the scaler (507)
increases the value to be divided, so the frequency of the DELAY CLOCK signal decreases and the
delay time of the delay circuit (506a) (506, b) The opposite operation to the above case is
performed such that In the comparator (512b), the optimum subsequent reverberation
reverberation time [Tsub] p of the multiplier (520) and the subsequent reverberation
reverberation time Tsub after the influence of the sound field from the encoder (511b) is applied.
When the subsequent reverberation reverberation time Tsub is larger than the optimum
subsequent reverberation reverberation time (Tsub) p, the counter CU sends the count-up signal
CU to the counter 508b. Content is converted to an analog value by a digital-analog converter
(509), passes through a smoothing circuit (514c), and is applied to the respective attenuators
(505a) (505b) of the reverberation circuits (504a) (504b), The signal causes the attenuation
value at attenuators (505a) (505b) to increase, so that the subsequent reverberation
reverberation time Tsub It operates so as that small.
However, if the subsequent reverberation reverberation time Tsub is smaller than the optimal
subsequent reverberation reverberation time (Tsub), the countdown signal CD is sent to the
counter (508b) and the content of the counter (508b) is digital. The analog value is converted to
an analog value by an analog converter (509) and passed through a smoothing circuit (514C) to
the attenuators (505a) (505b), which reduces the attenuation value in the attenuators (505a)
(505b) Therefore, the reverse operation to the above case is performed to increase the
reverberation time Tsub. By the way, the respective amplified signals before being added from
the pre-amplifiers (3A) (3r) are inputted to the interaural cross correlation coefficient calculator
(519), and the interaural cross correlation coefficient IACC Are analogized and input to the next
smoothing circuit (514b). Next, in the comparator (513b), the signal from the smoothing circuit
(514b) is compared with the target voltage signal from the variable resistor (515b) for setting
the target value of the interaural cross correlation coefficient IACC. If the signal from the
smoothing circuit (514b), ie, the signal corresponding to the binaural cross correlation coefficient
IACC, is larger than the target signal from the variable resistor (515b), the output from the
comparator (513b) The signal becomes smaller and is input to the input terminal att 1n (535) of
the sound field expansion device (516), and the sound field expansion device (516) operates so
as to make the interaural cross correlation coefficient IACC smaller. . However, if the signal
corresponding to the binaural cross correlation coefficient IACC is smaller than the target signal
08-05-2019
14
from the variable resistor (515b), the output signal from the comparator (513b) becomes larger
and the sound field As input to the input terminal att1n (535) of the expansion device, the sound
field expansion device (516) performs the opposite operation to the above case, such as
increasing the binaural cross-correlation coefficient LACC. By the way, the first reflected sound
delay time analyzer (510a) which receives the added signal added by the adder (503d) is a sound
pressure signal φ corresponding to the listening sound pressure. And this sound pressure signal
φ. Are input to the comparator (513a) after passing through the smoothing circuit (514a). In this
comparator (513a), the target voltage signal from the variable resistor (5I5a) for setting the
target value of the volume is compared with the signal from the smoothing circuit (514a). If the
output signal from the corresponding smoothing circuit (514a) is larger than the target value of
the volume from the variable resistor (515a), the output from the comparator (513a) becomes
larger, and the attenuator (505r) ( 505j7) acts to decrease the listening sound pressure.
Moreover, if the signal corresponding to the listening sound pressure is smaller than the target
value of the volume, the output from the comparator (513a) becomes smaller and the attenuation
of the attenuators (505r) (505β) becomes smaller. It works in the opposite way to the above
case to raise the listening sound pressure. The smoothing circuits (514a), (514b) and (514C)
appearing in FIG. 14 (a) are all capable of preventing the generation of noise due to a rapid signal
change. The delay times of the delay circuits (506a) and (506b) are also used to prevent abrupt
changes. Next, the operation of the sound field expansion device (516) will be described using
the sound field expansion device shown in FIG. 14 (b). First, the reverberation devices (504a)
(504b) to the input terminals 1nr (531) and 1njl! The difference component of the two signals
input to (532) is obtained by the adder (530a), passes through the low pass filter (537) through
the attenuator (536), and then the cross-talk phase is reversed. The phase from the input
terminal 1nr (531) to the output terminal outr (533) by the adders (530b) (530C) and the input
terminal inj! Output terminal out from (532) 1. (534) is added to each of the main routes up to
four. Apart from that, the output of the low pass filter (537) is further passed through the high
pass filter (538), and after the delay or phase shift rotation is given in the phase circuit or delay
circuit (539), Adders (530d) (530e) add to the respective main paths described above with a
phase such that the phase to be talked is in antiphase, and the signal after the addition is sent to
the output terminals outr (533) and outl C534). Each is output. The interaural cross correlation
coefficient IACC can be changed by changing the attenuator (536) of the sound field expanding
device (516). Further, this attenuator (536) is set so that the attenuation becomes larger as the
value of the signal from the input terminal att 1n (535), ie, the output signal from the comparator
(513b) becomes larger. Further, FIG. 14 (C) shows M, R, Schroeder (Schroeder) and B, S, Atal
(Atta's reverberator (504 c), and FIG. 14 (al) shows the reverberation apparatus shown in (al). If
this reverberator (504c) is used instead of (504a) and (504b), more natural reverberation can be
obtained.
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Although the microphone (2r) (2A) has been described to obtain the sound pressure at the
entrance to the ear canal of the dummy head or the human head (1) in the above embodiment,
the value of the inter-ear cross correlation coefficient IACC When it is not necessary to obtain V,
there may be one microphone, or even in the case of obtaining the interaural cross correlation
coefficient IACC, it may be performed by two microphones not attached to a dummy head or the
like. Although the audio signal is shown as a two-channel stereo, it is needless to say that the
two-channel signal may not be necessarily required. As described above, in the acoustic device
according to the present invention, the optimum first reflected sound delay time [Δt +) p is
measured from the acoustic signals received from the input terminals INR and INL and the input
terminals IN, and INL. (1) Optimum subsequent reverberation that outputs a reflected sound
delay time measurement means and an optimum subsequent reverberation reverberation time
(Tsub) l corresponding to (23 ± 10) times the optimum first reflected sound delay time [Δt +) p
The listening sound pressure φ in the sound field from the reverberation time output means and
the sound field signal from the microphone placed in the sound field. Listening sound pressure
measuring means for measuring the first reflected sound delay time measuring means for
measuring the first reflected sound delay time .DELTA.t1 from the sound field signal, and
measuring the subsequent reverberation reverberation time Tsub from the sound field signal The
following reverberation reverberation time measuring means, the interaural cross correlation
coefficient measuring means for measuring the interaural cross correlation coefficient IACC from
the sound field signal, and the above-mentioned optimum first reflected sound delay time [Δt +]
p and First comparison means for comparing the first reflected sound delay time Δt1 and
outputting a signal according to the difference, the optimal subsequent reverberation
reverberation time (Tsub) I, and the subsequent reverberation reverberation time Tsub Delay
time is changed by the output signal of the first comparison means and the second comparison
means that outputs a signal according to the difference, and the reverberation time is changed by
the output signal of the second comparison means. And input from the above input terminal
Reverberation means for adding reverberation to an acoustic signal, inter-aural cross-correlation
coefficient setting means capable of setting in advance the value of inter-aural cross-correlation
coefficient, and the above-mentioned inter-aural cross-correlation coefficient IACC The third
comparison means for comparing the setting value of the interaural cross correlation coefficient
setting means and outputting a signal according to the difference, and the output signal of the
reverberation means, receiving the third comparison Sound field setting means having an output
capable of changing the interaural cross correlation coefficient IACC of the sound field according
to the output signal of the means, and listening sound pressure setting capable of setting a target
listening sound pressure value in advance Means and listening sound pressure φ of the above
sound field.
And the comparison value of the listening sound pressure setting means, the fourth comparison
means for outputting a signal according to the difference, and the output of the sound field
expansion means as an input, the output of the fourth comparison means According to the signal,
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since the attenuation means capable of changing the attenuation factor, and the electroacoustic
conversion means for amplifying the signal from the above attenuation means and radiating the
acoustic signal into space are corrected, the sound field is corrected, and As well as being able to
create a preferred sound field, the first reflected sound delay time [Δt +] p and the optimum
reverberation time [Tsub] p which are optimal for the sound source in this acoustic device of the
present invention can be obtained, It is possible to change the value of the listening sound
pressure and the interaural cross correlation coefficient IACC in accordance with the preference
of the user. (2, + 2) Other acoustic devices related to the present invention Next, in a sound field
device in a sound field such as a passenger compartment with relatively short reflections (even
with extremely short reverberation time) By correcting the sound field based only on the three
physical parameters of the listening sound pressure in the sound field, the first reflected sound
delay time, and the subsequent reverberation reverberation time, a sufficiently more preferable
sound field can be created. Although it is needless to say that the acoustic device of the present
invention described above can be used as an acoustic device in the sound field targeted here, it is
possible to use such a small room, a relatively small reflection Japanese room or car. When
limited to a room, the other acoustic device related to the present invention can not only obtain
the same effect as the above-mentioned acoustic device of the present invention without using
the parameter of the interaural cross correlation coefficient IACC. Since the acoustic device is
compact and inexpensive, moreover, the circuit configuration is simpler than that of the abovedescribed acoustic device, so that the work efficiency is improved and the durability is improved.
Can be played. Therefore, one of the acoustic devices that can correct the sound field using the
three physical parameters of the listening sound pressure in the sound field, the first reflected
sound delay time and the subsequent reverberation reverberation time to create a more
preferable sound field An embodiment will be described with reference to the drawings. FIG. 15
is a block diagram of one embodiment of another acoustic device related to the present
invention. In the figure, (451) to (453) are absolute value circuits, (511 d) is an encoder, (551) is
an input terminal INR1 (552) an input terminal INt, and (553) is an integral reset signal input
terminal. (554) is a first reflected sound delay time analyzer, which is the circuit diagram of the
above-mentioned FIG. 13 (al and the first reflected sound delay time analyzer (510a) (510C) of
FIG. 14 (a); Although the internal configuration is slightly different from that of the time analyzer
(510b), the function is almost the same, and adding an attenuator (426) and a comparator (419)
to the configuration of a normal autocorrelator It is characterized.
Also in such a configuration, it operates in the same manner as the first reflected sound delay
time analyzer (510a) (510C) or the subsequent reverberation reverberation time analyzer (510b)
described above. The same reference numerals as in FIGS. 13 (a) and 14 (a) denote the same or
corresponding parts. First, when acoustic signals are input from the input terminals INII (551)
and INL (552), the two signals are added by the adder (503C) and input to the first reflected
sound delay time analyzer (544). Ru. In this first reflected sound delay time analyzer (554), the
same operation as the first reflected sound delay time analyzers (510a) (510c) shown in FIG. 14
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(a) is performed, and the output therefrom is the above-described first output. Similar to the case
of FIG. 14A, the encoder 511d is input to output the first reflected sound delay time [Δt +] p
which is optimum for the acoustic signal. A signal from the encoder (511d) is given to the scaler
(507) so as to obtain a delay time corresponding to the optimum first reflected sound delay time
[Δt +] I. The scaler (507) divides the CLOCK signal from the input terminal (521) to create a
DELAY CLOCK signal, and the DELAY CLOCK signal is sent to the delay circuits (506a, 506b) of
the reverberation circuits (504a, 504b). The acoustic signal is input from the input terminals INR
(551) and INL (552) to reverberation circuits (504a) and (504b), respectively, reverberant sound
is imparted, and sent to the attenuators (505r) (505j7). On the other hand, a signal φ0
corresponding to the sound pressure of the first reflected sound delay time analyzer (554) is
supplied to the comparator (513a) through the smoothing circuit (514a), and a variable resistor
for setting the target value of the sound pressure If the sound pressure is smaller than its target
value, compared with the voltage of the comparator (515a), the output of the comparator (513a)
becomes smaller, and the attenuator (505r) (505jl! ) Operate to reduce the amount of attenuation
by the signal sent from the comparator (513a). However, if the sound pressure is greater than its
target value, the output of the comparator (513a) will be high and the attenuator (505r) (505A)
will be attenuated by the signal sent from that comparator (513a) The opposite is done as in the
above case, such as increasing the amount. The output of the attenuator (505r) (505f) thus
obtained passes through the power amplifier (517r) (517I) and is reproduced by the speaker
(518r) (518β).
At this time, the optimum subsequent reverberation time (Tsub) I, which is the optimum value of
the reverberation time, is (23 ± 10) times the optimum first reflected sound delay time [Δt +) p
as described above. desirable. When a reverberator is configured in the form of reverberation
circuits (504a) and (504b), it is known that the subsequent residual phonological reverberation
time Tsub−one ε · Δt + / log (g) is obtained. . Here, g is an attenuation factor. Here, it can be
understood that the attenuation factor g should be in the range of 0.588 to 0.811, using the
relationship of Tsub = (23 ± 10) Δt1. When the subsequent reverberation reverberation time
Tsub = 23Δt1, the attenuation factor g = 0.741. That is, by setting the attenuation factor g in this
manner, the optimum first reflected sound reverberation time [Δt +) p and the optimum
subsequent reverberation reverberation time [Tsub3 p can be obtained. Of course, even if such
processing is applied, the influence in the sound field is added, but in the case of a space with
relatively little reflection such as a Japanese room or a space with extremely short reverberation
time even if there is much reflection such as a car cabin Is very effective. By the way, in this
acoustic device, in order to provide a simpler acoustic device, the same effects as those of the
above-described acoustic device or more can be obtained in a sound field such as a small room or
a relatively small reflection Japanese room or a passenger compartment. When this sound
system is used in a small room or a sound field other than the cabin, the performance is
somewhat lower than that of the above-mentioned sound system, but still the listening sound
pressure, the first reflected sound delay time and Since the sound field is corrected by the three
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physical parameters of the subsequent reverberation reverberation time, it is possible to create a
much more favorable sound field than conventional acoustic devices, and furthermore, this
acoustic device is the aforementioned acoustic device. The apparatus is more compact and less
expensive, has a simpler circuit configuration, and has the advantages of excellent workability
and durability. As described above, in this acoustic device, the optimum first reflection sound
delay time measuring means for measuring the optimum first reflection sound delay time from
the sound signals received from the input terminals INR, INL and the input terminals INR, INL
Listening sound pressure corresponding signal measuring means for measuring a signal
corresponding to listening sound pressure from the sound signal, and a signal corresponding to
the optimum first reflected sound delay time as a control signal, and inputting the sound signal, A
reverberation means in which a delay time corresponding to the optimum first reflected sound
delay time is added to the sound signal, and a subsequent reverberation reverberation time
thereof is set to (23 ± 10) times the optimum first reflected sound delay time And the listening
sound pressure setting means capable of setting the target listening sound pressure value in
advance, and the listening sound pressure output from the listening sound pressure
correspondence signal measuring means and the setting value of the listening sound pressure
setting means. , Comparing means for outputting a signal corresponding to the difference
between the two, and the output signal of the reverberation means, and changing the attenuation
factor given to the sound signal inputted from the reverberation means according to the output
signal of the comparing means Attenuating means, and an electro-acoustic conversion means for
amplifying the signal from the attenuating means and emitting an acoustic signal into space,
thereby correcting the sound field and creating a more preferable sound field. In this acoustic
device, the optimum first reflected sound delay time and the optimum subsequent reverberation
reverberation time can be obtained, and the listening sound pressure can be changed according
to the preference of the user.
As described above, according to the present invention, the acoustic characteristics of the sound
field that listens to music and the like are measured, and the listening sound pressure in the
sound field of the physical quantity and the psychological quantity, the first reflected sound delay
time. Subsequent reverberation reverberation time 1 Based on evaluating with parameters such
as interaural cross correlation coefficient, it is possible to provide an acoustic device that changes
its physical quantity to enable more preferable music listening by measuring this physical
quantity. The effect of being able to In addition, the output of the comparators 512, 419, and
420 is obtained in the above-described acoustic device, and the scaler (507) is configured to
create a DELAY CLOCK, which causes the first reflected sound to be delayed. Time can be
controlled. Configuring the DELAY CLOCK value to be able to be changed in response to various
music etc. by enabling it to be controlled manually from the outside is the case where Δt +: l p is
obtained in advance for various music etc. Is particularly useful and has the advantage of being
inexpensive to manufacture.
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[0002]
Brief description of the drawings
[0003]
Fig. 1 shows the relationship between the reflecting wall, the sound source and the human head,
Fig. 2 (al (b) shows an example of the normalized autocorrelation function measured using actual
music, Fig. 3 ( a) (b) shows the result of the preference test performed using music A and music
B, FIG. 4 shows the relationship between red and [Δt +) 17, FIG. 5 shows τp and (a) FIG. 6
shows the relationship between Tsub) p, FIG. 6 shows the relationship between normalized
preference and the value of IACC, and FIG. 7 shows the relationship between relative listening
sound pressure and preference measure S 1 8 shows the relationship between the delay time
between the direct sound and the first reflected sound and the measure S2 of the reference, and
FIG. 9 shows the relationship between the reverberation time of the subsequent reverberation
sound and the measure S3 of the preference. Figure 1O shows the relationship between IACC and
the measure S4 of the reference, Figure 1I FIG. 12 is a block diagram showing a schematic
configuration showing an embodiment of a sound field evaluation instrument related to the
present invention, and FIGS. Fig. 13 (c) (d) shows another example of the listening sound
pressure analyzer, Fig. 13 (Fig. 13 (d) shows the basic configuration of the reflected sound delay
time analyzer or the subsequent reverberation reverberation time analyzer. e) shows a
configuration example of a circuit for obtaining the interaural cross correlation coefficient, FIG.
14 (a) shows a configuration block diagram of an embodiment of an acoustic device according to
the present invention, and FIGS. Fig. 15 is a block diagram of a sound field expanding apparatus
and a reverberator, and Fig. 15 is a block diagram of one embodiment of another acoustic
apparatus related to the present invention.
In the figure, (1) is a human head or a dummy head, (2r) (2j ') is a microphone, (3H3r) (3f) is a
preamplifier, (4) is a physical quantity analyzer, and (5) is a comparator , (6) is a psychology
converter, (7) is a general evaluator, (8) is an output terminal, (9) is a recorder, αO) is a sound
field evaluation instrument, CD is a listening sound pressure analyzer, IAZr (510a) (510c) (554)
is a first reflected sound delay time analyzer, U (510b) is a subsequent reverberation
reverberation time analyzer, (a) is an interaural cross correlation function analyzer, and S1 is a
listening sound Pressure psychology converter, ■: first reflection sound delay time psychology
converter, ■: subsequent reverberation reverberation time psychology converter, figure:
interaural cross correlation coefficient psychology converter, (451) ~ ( 453) is an absolute value
conversion circuit, (501) (502) (521) (531 (532) (535) (5) 1) (552) is an input terminal, (503a)
to (503d) is an adder, (504a) (504b) is a reverberator, (505a) (505b) (505r) (505β) is an
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attenuator, (506a) (505a) 506b) is a delay circuit, (507) is a scaler, (508a), (508b) is an up /
down counter, (509) is a digital-analog converter, (511a) to (511d) is an encoder, (512a) (512b)
513b) (513b) is a comparator, (514a) to (514C) is a smoothing circuit, (515a) (515b) is a
variable resistor, (516) is a sound field expanding device, (517r) (51M) is a power amplifier,
(518r) (518f) is a speaker, (519) is an interaural cross-correlation coefficient calculator, (520) is
a multiplier, (533) (534) is an output. It is a terminal. In the drawings, the same reference
numerals denote the same or corresponding parts. Figure 1
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