JPH039226A - Acoustic device - Google Patents

Abstract

PURPOSE:To vary physical quantities and to enable preferable music listening by evaluating the acoustic characteristics of a sound field from variables and constants of listening sound pressure, 1st delay time, etc., of the sound field which are physical quantities and mental quantities and measuring the physical quantities. CONSTITUTION:A dummy head or human head 1 is installed in a place to be evaluated, microphones 2l and 2r fitted in the left and right external auditory meatuses detect the sound pressure signals of both ears, and a preamplifier 3 amplifies and adds the signals. Then a physical analyzer 4 finds the physical quantities, i.e. the 1st best reflected sound delay time, the best following reverberation sound reverberation time for the time which is (23+ or -10) times as long as the 1st best reflected sound delay time, and the cross correlation coefficient IACC between both the ears from the sound pressure signals of both the ears. The respective values are compared by a comparator 5 with the respective best values which are inputted as comparison data from an input terminal (y) and converted by a metal quantity converter 6 into quantities corresponding to metal quantities; and a total evaluation device 7 performs total evaluation. Consequently, the preference of the sound field is outputted from an output terminal 8.

Description

[Detailed Description of the Invention] This invention focuses on converting the acoustic characteristics of the sound field for listening to music into the form of physical and psychological quantities that are easy to evaluate, and changes the physical quantities so as to enable more preferable music listening. The present invention relates to an audio device that can perform

Research has been conducted on desirable sounds and sound fields, but the research conducted this time revealed that the following parameters are necessary to determine a good sound field in a concert hall, which is the standard for the sound reproduced by audio equipment. It gradually became clear that there was something to show.

In other words, there are four important objective parameters that represent the characteristics of binaural sound: audible sound pressure, delay time of the first reflected sound, reverberation time of subsequent reverberant sound, and interaural cross-correlation coefficient. That is, the breference (
This was revealed through tests of auditory comfort.

Next, the above four elements will be explained in detail.

First, Figure 1 shows the relationship between a sound source and a person's head in a space with reflective walls. In the figure, αυ is the human head, α force is the sound source, and 030 is the reflecting wall. Here, the sound source signal is p(t
), and when the impulse responses from the sound source to the left ear and right ear are hp(t) and h r(t), respectively, the left and right ear signals fz(t) and f, (D are respectively fz( t)=fo“p (ν)h仝(t-ν)dν=p
(t)*l-12(1) (Formula 1a) f, (t) = fo'p(L/) h, (t-L/)
It is expressed as (t) * h , (t) ... (21b). The * mark in the above formula indicates convolution.

In the figure, n = 0 indicates the case where the beam enters the ear directly.
n = 1 means that the sound reflected by the reflective wall αQ enters the ear, and n
=2 means that the sound reflected by the reflective wall α fathom enters the ear, m=
1 indicates the case where the sound enters the left ear, and m=r indicates the case where the sound enters the right ear.

By the way, in Fig. 1, the reflecting wall a': a C14)
The number of sounds entering the ear after being reflected by the wall is shown only two for each reflecting wall, but assuming that many such reflections occur, the impulse response upon reflection at this reflecting wall α30 is expressed as W n (t ) Then, the impulse responses reaching the left ear and the right ear (t), h, (t) are respectively hp(t) = Σ G, W, , (−△t, )*
h, 1 (t)n=0 ... (Formula 2a) h, (t) = Σ G, W, (-△t, )
*h, , (t)n=0 (Formula 2b). Using these (Formula 2a) and (Formula 2b), the above (Formula 1a) and (Formula 1b) become fz
(t)=Σ p(t)*G,,W,,(t-
△t,)*h,(t)n=0...(Formula 3a) f r(t)=Σ p(t)*G nW, (t-
Δt , )* h , 1 (t)n=0 (Formula 3b).

Here, if the sound source p (t) does not have −-like radiation characteristics, p (t
) to p. (r) can be substituted.

By the way, the information in the acoustic signal that enters both ears is
It includes independent and objective acoustic parameters, the first of which is the sound source signal p(1). Using the sound source signal, the long-term autocorrelation function Φp(τ) can be expressed as (Equation 4). where put) −p(t)
*s (t), where s (t) corresponds to the sensitivity of the ear,
Theoretically, it is expressed by the characteristics of the middle ear and the outer ear, but in practice it can be expressed as the impulse response of a G filter, which is well known as an approximation of auditory characteristics. Naturally, by dividing (Equation 4) by the power Φp(0) of p=(t), the normalized autocorrelation function φp(τ) can be expressed as Φp(o).

FIGS. 2(a) and 2(b) illustrate 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 will be referred to as music A hereinafter. Figure 2(b) shows Arnold
) music by “Sinfornietta 9 pieces 48: ■ Movement, Allegro con brio (Synfornietta
, 0pus48: m Movement, A
This music is hereinafter referred to as music B.

Next, the second objective parameter is the impulse response caused by reflection at boundaries such as walls. This is related to the initial time delay between the direct sound and the first reflected sound, as well as to the initial reflected sound, subsequent reverberant sound, and changes in the spectrum due to reflection.

The third objective parameters include impulse responses to the left and right ears, z(t), h, , (t). These impulse responses play an important role in sound localization, and are not independent of each other. This means that for a centrally localized signal, h, , z(t), and h. , (t).

Next, take these two impulse responses h , , v (t). , (t).

First, the long-term binaural cross-correlation function 09. between the binaural signals fz(t), f, (t). (τ) as f or (t), f
, (t), τ1≦Ims... (Equation 6) By the way, the feeling of spaciousness or sense of lack of direction occurs when the interaural cross-correlation is a small value, and it is determined that In the case of only signals from directions, the binaural cross-correlation function has a large peak when 1τ1<1ms. where 1 and 1<1
ms is the binaural signal fv(t), f r
This is because the time difference between (t) cannot normally be more than 1 ms due to the relationship between the interaural distance and the speed of sound.

First, the binaural cross-correlation function Φ and (τ) due to direct sound only
is fx (t) = p(t) *h, t(t).

As fr(t)=p(t)*h,,(t)(
It can be obtained by substituting into equation 6). By the way, the normalized autocorrelation function φ cross, (τ) of only the direct sound can be expressed as... (Equation 7). Furthermore, φ sentence, (τ)°h to him,
If v(t)=h,,(t), then it becomes approximately l.

Here, Φ\"x(o) and Φ,';(o) indicate the autocorrelation function of τ-0 of the signal at each of the left and right ears.

Next, if another reflected sound is added to the direct sound after the time when the autocorrelation function of the direct wave becomes small, then the normalized autocorrelation function φ2. (τ) is expressed as follows (Equation 8) when W and (t) are equal to Dirac's delta function δ(1). Here, Φ and (τ) are the interaural cross-correlation functions of the n-th reflection, Φn"ff1(o), Φr:'(0
) indicates the autocorrelation function at τ=0 of the n-th reflected sound of each left and right ear. Note that when the sound source is in front of the user in a normal room, the maximum value of the interaural cross-correlation function is obtained extremely close to τ=0.

By the way, let IAAC be the strength of the normalized interaural cross-correlation, IAAC = 1 φg, (r)l max fo
It is defined as rl r l ≦ lm5 (Formula 9). In addition, φ2. (τ) = Φ cross, (τ) 7℃1
1. This is a Y edition, and Φ(0), Φ2, and (0) respectively indicate the autocorrelation function at τ=0 of the signal sound for the left and right ears.

Above, we have discussed the long-term autocorrelation function of the sound source signal, the multiple impulse responses caused by sound reflecting off walls, etc., and the interaural cross-correlation function that shows the correlation between binaural signals. , then from these physical quantities,
Four elements are important objective parameters that represent the characteristics of binaural sound signals: listening sound pressure, delay time of the first reflected sound, reverberation time of the subsequent reverberant sound, and interaural correlation function. We will explain how to find the optimal value for each value.

The results of the preference test conducted using the aforementioned music A and music B are shown in FIGS. 3(a) and 3(b). The horizontal axis is IACC1, and the vertical axis is audible sound pressure (unit: dBA). The audible 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 the preference. Figure 3 (a) (Clearly from bl, the optimal listening sound pressure [p) p does not depend much on IACC; it is 77 to 79 dBA for music A with a rather slow tempo, and 77 to 79 dBA for music B with a fast tempo. It can be seen that it is 79 to 80 dBA in some cases. In either case, the optimal value for hearing sound pressure is 79 dBA.
It can be seen that the values before and after are taken.

Next, according to the results of Breference evaluation of a synthetic sound field consisting of a direct sound and a single reflected sound when reproduced by a speaker using music and speech, the normalized autocorrelation function φp(τ) of the sound source signal is Find the level of the reflected sound as ±6d of the direct sound.
B, the optimal delay time of the reflected sound is 1Φp(τ) equal to l/10 of the level G1 of the first reflected sound.
It turns out that the time corresponds to that corresponding to .

Therefore, Φp(τ)1 is 1/1 of the level G1 of the first reflected sound.
FIG. 4 is a diagram in which the horizontal axis represents the time τd corresponding to 0, and the vertical axis represents the delay time [Δt+)I) of a single reflected sound at which the breference is maximum. The range shown in the figure shows the delay time when it is 0.1 lower than the maximum value of the reference, and the symbol O in the figure is the level of the first reflected sound G, = 6 dB, ・ is G1 = OdB, mouth is GI=-6d
Each time of B is shown. Especially 1φp(τ)1
If τp is the time when φp(0) becomes 0.1 times, then Gl
=OdB, it can be expressed as τd=τp. Incidentally, the τp of the aforementioned musics A and B are 127 m5 and 35 m5, respectively, as can be inferred from FIGS. 2(a) and (b). As is clear from Figure 4, a straight line can be drawn to approximate each point in the diagram, and when looking at both axes corresponding to the straight line, τd is the single line that maximizes the breference. It can be seen that the delay time [Δt+) of reflected sound almost coincides with p. At the same time, it almost coincides with the time τρ at which 1Φp (τ)1 becomes 0.1 times Φp(0). That is, if this is expressed mathematically, the most preferable delay time [△t1]p of the first reflected sound is [△t,]p=τp ・(Formula 10%Formula% Φp (τ)1≦KG' ・・・(Formula 11)τ
〉τp, 1φp(τ)1≦0.1... (Formula 1
2) It can be expressed as.

Furthermore, the autocorrelation function of the sound source signal has a close relationship with the optimal reverberation time, and this relationship is shown in FIG. The vertical axis is the optimal reverberation time (Tsub) p of the subsequent reverberant sound, and the horizontal axis is τp. The reverberation time mentioned here is 60 d of direct sound.
It is expressed as the time it takes for the signal in the reverberation section to attenuate by 60 dB, rather than the time it takes for the signal in the reverberant section to attenuate by 60 dB.

In the figure, Music A and Music B are the same as those mentioned above, but Music E is Mozart's "Symphony in C major, 551 Symphony, 4th movement, Molto Allegro", and Speech S is Doppo Kunikida's "The Sound of Tone's Water"[You'll be surprised at how blue the sky is, how bright and high it is. J (τp=12ms). As is clear from the figure, the relationship shown in FIG.
It can be approximately approximated by a function of (23±10)rp.

Next, Figure 6 graphically represents the results measured in a synthetic sound field consisting of a direct sound and a single reflected sound, with the horizontal axis representing the arrival direction ξ of the reflected sound and the vertical axis representing the normalized The reference and IACC values are shown. It can be seen from FIG. 6 that the preference increases as the value of IACC decreases. That is, the Brefference score and IA
The correlation function with the value of CC is negative (-0,76: 1% significance level), and this holds true when IACC takes the maximum value at τ=0. most effectively IC
It can also be read from the figure that in order to reduce AA, the early reflected sound should arrive within a range of ±(55°±20°) from the front.

Next, a measure of preference based on the above-mentioned four parameters will be described. By the way, this scale of preference was determined through a comparative test, and each parameter independently influences the scale of preference. As a result, since the principle of superposition can be applied, by normalizing each objective parameter by the optimal value, it becomes possible to generalize the sound field breference data obtained in a concert hall or the like.

The preference scale obtained from the comparative test will be explained below.

First, the measure of brefference S as a function of hearing sound pressure
1 is shown in FIG. In this figure, the scale at optimal hearing sound pressure is set to zero.

The value of Sl is approximately symmetrical with respect to the most preferable listening sound pressure (OdB), but it is slightly more symmetrical when it is shifted towards the weaker listening sound pressure than when it is shifted towards the stronger sound pressure. Breference scales tend to be good.

Expressing this mathematically, 5t(LL)≦S+(LL) (Formula 13)%
The formula is %)), where p is the audible sound pressure, and (p)p is the optimal audible sound pressure. Note that in the figure, ○ indicates the value of music A, and X indicates the value of music B.

Next, a measure S2 of breference as a function of the delay time between the direct sound and the first reflected sound (first reflected sound delay time) is shown in FIG. The horizontal axis of this figure is the most preferable time delay [
It is normalized by Δt+)p. By the way, the second
The delay time of reflected sound is the optimum value [△t,] p is [△t
Since it is known that z ]p zl, 8 rp, this optimum value was used as the delay time of the second reflected sound in the measurement. Of course, it goes without saying that the condition of (Equation 10) exists. In the figure, ○, a, A
Each of △ is an individual measurement result with music A, and
, b, and B△ are individual measurement results for music B, C for music C, D for music D, mouth for music E, and * for speech S.

Music A, B, and E have already been mentioned earlier, but music C
is Hyton's Symphony No. 102 in flat major, second movement Adagio (τp = 65ms), and the music is Wagner's Siegfried Idyl.
l) 332 bars (r p =40 ms). Note that, as in FIG. 6, the optimal measure of the delay time of the first reflected sound is set to zero.

Furthermore, it is a measure of preference S as a function of the reverberation time of the subsequent reverberant sound, and the dashed line is that for G-1,l. In the figure, ○, a is for music A,
, b is the experimental result for music B, mouth is for music E, and ・ is the experimental result for speech S, all of which are G.
=4.1. G=4.1 corresponds to a case where there is a lot of reverberant sound, such as when being at the rear of a concert hall, and G=1.1 corresponds to a case where there is a lot of direct sound, such as at the front seats of a hall. The preference scale at the most preferred reverberation time is set to zero.

Next, FIG. 10 shows the preference measure S as a function of IAcc. In the diagram, ○ is for music, X
, b are the experimental results for music B.

Due to the nature of IACC, if the maximum value is taken at τ = 0,
The sound image is in the front direction. As is clear from the figure, l5C
As C approaches 1, the preference measure S4 decreases rapidly. Therefore, it is necessary to make IACC smaller than 0.5 as much as possible.

So far, there are 4 scales of reference S! to S4, each of these four scales can be approximated by the approximate expressions shown below.

First, the hearing sound pressure breference scale St shown in FIG. 7 is S+=-α11X+l"'...
) However, X+ = 201og (p/ (p) p) −(
Equation 15) % Equation % The preference measure S2 is S2 = −α21 X21 ”” However, X2 = log (Δt1/[Δt+]p). Note that (Formula 17) ... (Formula 18).

Furthermore, the preference scale S3 for the reverberation time of the subsequent reverberant sound shown in FIG.
) However, X3 = log [Tsub/ l:Tsub) p)
- (Formula 21) %Formula % Next, the delay time of the first reflected sound shown in Fig. 8 is... (Formula 22) If α3 is negative in (Formula 22), then α3
=0... (Formula 22').

The remaining IACC brilliance measure S4 shown in FIG. 10 is 54=-α4
) However, X < = IACC...(Formula 24)%Formula% α4=1.45±0.44...(
Equation 25).

Based on the principle of superposition, the overall measure S of preference in concert halls etc. is S = Σ 5 i = 1 ... (Equation 26) The % formula % (1 2, 3, 4) is the above-mentioned four blocks. Shows reference scale.

Brefference measures S and S obtained in this way
+ (i=L 2, 3.4) allows accurate evaluation of the sound field.

This invention uses four parameters to measure the preference in a concert hall or room: the audible sound pressure in the sound field, the delay time of the first reflected sound, the reverberation time of the subsequent reverberant sound, and the interaural cross-correlation coefficient. The present invention focuses on the fact that it is possible to obtain the respective brefference scales and the overall brefference scale, and provides an acoustic device that can create a more preferable sound field by correcting the sound field based on each parameter. That is the purpose.

(1) Sound field evaluation measuring device related to the present invention First, a sound field evaluating measuring device related to the present invention will be described using one embodiment.

FIG. 11 and FIG. 12 are block diagrams showing the general configuration of the sound field evaluation measuring instrument. FIG. 12 is a block diagram showing each of the four parameters. In the figure, (1) is a human head or dummy head, (21) (2r) is the external ear, a microphone inserted into the entrance, (3) (3j?
) (3r) is a preamplifier, (4) is a physical quantity analyzer, (
5) is a comparator, (6) is a psychological quantity converter, (7) is a comprehensive evaluator, (8) is an output terminal of the blind reference, (9) is a recorder for recording the blind reference, and Oo) is a Sound field evaluation measuring instrument, example is adder, riυ is listening sound pressure analyzer, (transformation)
is the first reflected sound delay time analyzer, (c) is the subsequent reverberation sound reverberation time analyzer, (b) is the interaural cross-correlation function analyzer, GD
is a hearing sound pressure psychological quantity converter, ■ is a first reflected sound delay time psychological quantity converter, ω is a subsequent reverberant sound reverberation time psychological quantity converter, the figure is a binaural cross-correlation coefficient psychological quantity converter, (y )(y+) (
Y2), (Y3), and (Y4) are comparison data input terminals.

The comparison data input terminal (y) (y+) (yz) (yz
) (y4) have their respective optimal values input, and the input terminal (yl) receives a signal corresponding to the optimal hearing sound pressure ((p) p ), and the input terminal (y2) receives a signal corresponding to the optimal first reflection. A signal corresponding to the sound delay time ([△t+)l]) is sent from the input terminal (y3) to the last subsequent reverberation sound reverberation time (CTsub
) p) A signal corresponding to the optimal interaural cross-correlation coefficient is input from the input terminal (y4).

In addition, the listening sound pressure analyzer CD, first reflected sound delay time analyzer, subsequent reverberation sound reverberation time analyzer (c), and interaural cross-correlation function analyzer shown in Figure 12 are the physical quantity analysis in Figure 11. Vessel (4
) corresponds to the physical quantity analysis section. Hearing sound pressure psychological quantity converter Gυ in Figure 12, first reflected sound delay time psychological quantity converter ■, subsequent reverberation sound reverberation time psychological quantity converter Am, interaural cross-correlation coefficient psychological quantity converter (support) is the comparator (5) in Figure 11.
and a psychological quantity converter (6). Similarly, the comprehensive evaluator (7) corresponds to the comprehensive evaluation section.

First, a simple flow of one embodiment of this sound field evaluation measuring instrument will be explained using FIG. 11. First, a dummy head or a human head (1) is installed at a location such as a concert hall or listening room where a sound field evaluation is desired. Microphones (2β) (2r) attached to the left and right entrances of the ear canal
The sound pressure signal of both ears is detected by the preamplifier (3).
After amplification and addition, the physical quantity analyzer (4) converts the sound pressure signals of both ears into physical quantities, that is, hearing sound pressure, first reflected sound delay time, subsequent reverberant sound reverberation time, and binaural cross-correlation coefficient. Find IACC and compare each value with a comparator (5
) is compared with each optimal value input as comparison data from the input terminal (y), and then compared with the respective optimal values input as comparison data from the input terminal (y), and then
It is converted into a psychological quantity, that is, a quantity corresponding to the preference, by a psychological quantity converter (6) that performs processing based on Formula 14), (Formula 17), (Formula 20), and (Formula 23), and then in a comprehensive evaluator (7). The overall evaluation is performed and the breference in the sound field is outputted from the output terminal (8). This calculator can be used to evaluate the sound field of concert halls and listening rooms.

Next, the embodiment shown in FIG. 11 will be explained in detail with reference to FIG. 12, which shows the flow of four individual parameters.

As mentioned above, the sound pressure signals detected by the microphones (1) (2r) are amplified by the preamplifier (3f (3r)), added by the adder!4■, and then sent to the physical quantity analysis section. The corresponding audible sound pressure analyzer CD, the first sound ejection sound delay analyzer, and the subsequent reverberation sound reverberation time analyzer are inputted to each analyzer, and the audible sound pressure p and the first sound ejection sound pressure are input to each of the corresponding analyzers. Each of the sound delay time j!+ subsequent reverberant sound reverberation time Tsub is analyzed and measured and output.In addition, apart from these three flows, two preamplifiers are added before being added by the adder Q@. (31) The output from (3r) is input to the interaural correlation function analyzer (transfer) corresponding to the physical quantity analysis section, where the maximum value IACC of the interaural cross-correlation coefficient is analyzed and output. Next, there are four analyzers (41 (transformation)) corresponding to the physical quantity analysis section.
The hearing sound pressure p, the first ejection sound delay time △jl+, the subsequent reverberant sound reverberation time Tsub, and the maximum interaural cross-correlation coefficient IACC output from each of It is input to the corresponding listening sound pressure psychological quantity converter 61), the first reflected sound delay time psychological quantity converter port, the subsequent residual sound is input to the reverberation time psychological quantity converter group, and the other comparison data input terminal (
Optimum hearing sound pressure [:p) p, optimal first reflected sound delay time [△t+31), optimal subsequent reverberation sound reverberation time [Tsub] input from each of y+)(yz)(y3)(y4)
p is compared with the optimal interaural cross-correlation coefficient.

The optimal binaural cross-correlation coefficient may be considered to be 0, but in practice there are many cases where there is no problem in setting it to 0.4 to 0.5 or less, so in this example, it is set to 0.4. ing.

The calculation formulas for this comparison and psychological quantity conversion in the psychological quantity converter (51) are the aforementioned (Equations 14) to (Formula I6), the first reflected sound delay time psychological quantity, and In the converter ω, the above-mentioned (Equation 17) to (Equation 19),
In the subsequent reverberation sound reverberation time psychological quantity converter■, the above-mentioned (
Equations 20] to (Equations 22-), interaural cross-correlation coefficients in the psychological quantity converter (2) are as shown in (Equations 23) to (Equations 25). Brefference measures S1, S2 and S3, which are the outputs
and S4 can be determined by calculation processing using a microcomputer program or the like. However, if accuracy is not a big issue, a conversion table as shown below may be used.

Conversion table ■ LL (dB) 10, 7 10, 5 10, 3 10, 2 10, IO20 -0,1 -0,2 10, 3 10, 5 10, 8 -1 .2 LL=2010g(p/ [1):] p) Conversion table■ △1+/△t+)p 2 0.1 0.2 0.3 0.4 0.5 0.6 0.7 0.8 0.9 1.0 2.0 3.0 4, 0 5.0 6.0 7.0 8.0 9.0 10.0 -1.1 10, 6 10, 4 10, 2 1 〇, l -〇, 0 -0.0 -0.0 10, 0 0, 〇10, 2 10, 4 -0.7 -0.8 1, 0 -1.2 -1.3 - 1.4 -1.5 The above conversion tables ■ and ■ are respectively audible sound pressure psychological quantity converters 1! This is an example of a conversion table between in and the first reflected sound delay time psychological quantity converter diagram. These conversion tables I and II include the hearing sound pressure p, the optimal hearing sound pressure (pop), the respective breference scales S+ 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 for the audible sound pressure psychological quantity converter (51) and the first reflected sound delay time psychological quantity converter ω is shown here, the following reverberation sound reverberation time psychological quantity converter ω is stored. A similar conversion table can be created for the binaural cross-correlation numerical psychological quantity converter diagram.

In this way, the converter (51) of the comparison and psychological quantity converter
The Breference scales 81 to S4 output from ~
The overall preference measure S obtained as a result of the comprehensive evaluation is input to the comprehensive evaluation device (7) and output to the output terminal (8). This output is recorded on the recorder (9). However, in this embodiment, the recorder (
9), but it does not necessarily have to be included, and it goes without saying that it may also be replaced with a display.

Next, the audible sound pressure analyzer α0. of the physical quantity analysis section used in the sound field evaluation measuring instrument 00) in FIG. Figures 13 (a) to (e) are shown for an example of the configuration of the first reflected sound delay time analyzer, the subsequent reverberant sound reverberation time analyzer (2), and the interaural cross-correlation function analyzer (foundation). I will explain using

Figure 13 (al is the first reflected sound delay time analyzer using a square integral type reverberation meter by M, R, 5 chroeder, which is a modified correlator, or the subsequent reverberant sound reverberation time analysis. This is a diagram showing the basic configuration of Kisho, which has the function of an audible sound pressure analyzer that also includes output corresponding to sound pressure.Figure 13 tb+
is a diagram showing 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 □□□ or the subsequent reverberant sound reverberation time analyzer. . Figure 13 (C) and (
d) is a diagram showing another example of the configuration of the audible sound pressure analyzer.

FIG. 13(e) is a diagram showing an example of the configuration of a circuit for calculating the interaural cross-correlation coefficient.

In FIG. 13(a), (400) is the input terminal of a circuit used as a first reflected sound delay time analyzer or a subsequent reverberant sound reverberation time analyzer using a square integral type reverberation meter;
401) (402) (403) are delay circuits, (404)
- (411) are multipliers, (412) - (4]5) are integrators, (416) - (418) are adders, (419) - (
421) is a comparator, and (422) is an autocorrelation coefficient φ. The output terminals (423) to (425) are 01 to 0, respectively, which output information on the delay time of the first reflected sound or the reverberation time of the subsequent reverberant sound.

(426) is the output terminal of the attenuator (Tb) in Fig. 13, (430) is the output terminal (423) to (425)
01~o+i-1 outputted from the input priority encoder, X1~Xn-+ are code outputs corresponding to the first reflected sound delay time or the subsequent reverberant sound reverberation time,
In Fig. 13(C) and (d), (435) is a rectifier circuit, (436) is a reduced pass filter, (437) is a multiplier, and in Fig. 13(e), (441) is a binaural mutual phase The input terminal where the left input signal Lin of the circuit for calculating the relational coefficient is input, (442) is the same (the input terminal where the right input signal Rin is input, (443), (444) is the adder, (445)
(448) is a root mean square circuit (RMS), (449) is an arithmetic circuit, and (450) is an output terminal for taking out an output corresponding to the cross-correlation coefficient.

Note that in the circuit of FIG. 13(a), there are (n-1) delay circuits, adders, and comparators each, and a multiplier to which the output signal from the delay circuit is input, an integrator, and an output signal from the integrator. is equipped with n input multipliers and n output terminals, but in the drawing, (n-1) multipliers are represented by 3
Only four representative numbers are drawn, and the remaining ones are omitted, and consecutive symbols are assigned to those drawn in the figure. Therefore, the second delay circuit (402) is followed by the (n-1)th delay circuit (403), and the third to (n-2)th delay circuits are not labeled for convenience.

First, FIG. 13 [a] will be explained. Input terminal (4
When a signal is given to the multiplier (00), the input signal is sent to the multiplier (
404) to (407), and the delay circuit (40
1) is also given. The output of the delay circuit (401) is given to the multiplier (405) and the output of the next delay circuit (40
2) and sequentially delayed signals are produced up to the output of the (n-1)th delay circuit (403). The output of each delay circuit (401) to (402) is sent to a multiplier (405).
) to (407). Multiplier (404)
performs a square operation on the input signal itself. Next, the output signals of the multipliers (404) to (407) are integrated in integrators (412) to (415), respectively, and the output signals are further squared by multipliers (408) to (411). In addition, the output signal of the integrator (412) is directly input to the terminal (422) in addition to the input to the multiplier (408).
is output to. This signal has an 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 squaring the power can also have an equivalent function simply by converting it into an absolute value. By the way, the output signals of the multipliers (40g) to (411) are the square of the autocorrelation coefficient φ. 2φ12. ......, φ2o
-1 signal is added to adders (416) to (418), the outputs of adders (416) to (417) are respectively added to the next adder, and the (n-1)th adder (418) )
It will be carried out until By the way, the output signal from the multiplier (408) and the output signal from the adders (416) to (417) are output from the comparison circuits (419) to (421), respectively.
At , the output signal of the last adder (418) is compared with the signal after being attenuated by the attenuator (426), and the respective output signals 01 to 01 are sent to the output terminals (423) to (425).
(423) to On-+ (425) are output. The attenuation ratio of the attenuator (426) is set to (0,1)' when determining the first reflected sound delay time, and to (0,001)' when determining the subsequent reverberation sound reverberation time. is used. This is similar to what is already known as an optimal delay time detector, and the value of the autocorrelation function is 1710.
Alternatively, the delay time is determined to be 1/1000. The total delay time from the delay circuit (401) to the delay circuit (403) is about the length of the first reflected sound delay time and the subsequent reverberation sound reverberation time, that is, 100 to 200 respectively.
m5 and about 3 to 5 SeC are required. Naturally, as this time elapses, the integration circuits (412) to (41
5) needs to be reset. However, the reset circuit is not shown. Next, the priority encoder (430) shown in FIG. 13(b) is
Output signals of comparators (419) to (421) shown in ○,
(423) to ○, , -, (425), and the attenuation ratio (
Display or output the code X1...Xn corresponding to the delay time τ (-τp) (the delay time that first becomes smaller than the damping ratio) corresponding to 0,1)' or (0,001)2. .

The following Figures 13 (C) and (d) are diagrams showing other examples of circuits used as audible sound pressure analyzers;
3. Output signal φ from the output terminal (422) in FIG. 3(a). You can find out the audible sound pressure in the same way as you can. This configuration uses a low-pass filter (43
6) and a rectifier circuit (435) or a square multiplier (437).

The circuit (519) for determining the interaural cross-correlation coefficient shown in FIG. 13(e) uses an example of a circuit system for determining the 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 signal L in,
Rin is a direct mean square circuit (445).

(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 from an adder (443), and the difference signal is taken out from an adder (444) which makes one signal negative and adds it to the other signal, and the root mean square circuit (446) takes out the sum signal between the input signals Lin and Rin. ) and (447), and similarly output the sum signal and the difference signal as their respective root mean square values α and β. The four root mean square signals α, β, γ, and δ thus obtained are subjected to arithmetic processing (α2β2)/(4γδ) in an arithmetic circuit (449), and the mutual correlation is output to an output terminal (450). Output φ corresponding to the number
(In other words, the interaural correlation coefficient (IACC) is obtained. However, when using the circuit shown in FIG. 13(e), input signals Lin and Rin are connected to input terminals (441) (442).
If a delay circuit of 0 to 1 ms is provided before the input to the input signal, and the delay time is varied to find the maximum value, the circuit configuration corresponds to the above-mentioned (Equation 9), and from that circuit, (Equation 9) ) is required. The delay circuit for this purpose is shown in Figure 13 (not shown in el).The more accurate interaural correlation coefficient (IACC)
In order to perform the calculation, it is preferable to perform the numerical calculation using the above-mentioned formulas (Formula 7) to (Formula 9).

In the embodiment described so far, the inputs to the audible sound pressure analyzer CD, the first sound ejection sound delay analyzer, and the subsequent reverberant sound reverberation time analyzer Sho are the dummy head or human head ( In 1), we have used the sum signal of the two acoustic signals entering the left and right ears, but the delay time between the two signals is usually less than 1 ms, and the analysis by these three analyzers Aυ~ has been used. Since there is no significant difference in the results, sufficient analysis results can be obtained even if one of the left and right acoustic signals is used. The optimal listening sound pressure [p) of the sound field evaluation measuring instrument 00) which is an embodiment of the first invention described above varies depending on the type of sound source, but it is approximately (79±5) Since it is about dB, 79 dB is used here as a representative. In addition, the optimal first reflected sound delay time [△t+3+] and the optimal subsequent reverberation sound reverberation time [Tsub
]p is the first reflected sound delay time analyzer from the actual sound source itself to determine the delay time τp that first becomes smaller than the attenuation ratio, and then calculates this as the optimal first reflected sound delay time [△t1]
It is preferable to set it to p, and multiply it by (23±10) and use it as the optimal subsequent reverberation sound reverberation time [Tsub) p.

In this example, (Tsub) p =23rp is used.

As mentioned above, according to this sound field evaluation measuring instrument, one or two
a microphone, an amplifier for amplifying the acoustic signal obtained from the microphone, and an audible sound pressure measuring means for measuring the audible sound pressure p from the amplified signal; A first reflected sound delay time measurement time for measuring the first reflected sound delay time Δt1 from the signal, and a subsequent reverberation sound reverberation time measurement means for similarly measuring the subsequent reverberation sound reverberation time Tsub from the amplified signal. , furthermore, a binaural cross-correlation coefficient measuring means for measuring the binaural cross-correlation coefficient IACC from the amplified signal, and a ratio of 10 of the audible sound pressure p and the optimum audible sound pressure [p)p. The value of the absolute value of the 20 times the logarithm to the 372nd power with 04±0.02) is outputted as a scale S1 of Brefference (psychological quality in terms of hearing). In other words, the first comparison conversion means outputs the output corresponding to the above (Equation 14) to (Equation 16), the first reflected sound delay time Δt1 and the optimal first reflected sound delay time [Δt,).
When the first reflected sound delay time is greater than the optimal first reflected sound delay time, the value is the 3/2 power of the absolute value of the logarithm to the base 10 of the ratio to p, -(1,42±0,6 ] is multiplied by -(1,11±0.5) when it is small, and outputs the value obtained as the preference scale S2, that is, outputs corresponding to the above-mentioned (Equations 17) to (Equations 19) are output. A second comparison conversion means sets the subsequent reverberation sound to the 372nd power of the absolute value of the logarithm to the base 10 of the ratio between the subsequent reverberation time Tsub and the optimal subsequent reverberation sound reverberation time (Tsub) p. When the reverberation time is larger than the optimal subsequent reverberation sound reverberation time, the sum of -(0,74±0.25) times the sound pressure of the entire reflected sound and -(0,45±0.15) is If it is small, multiply the total sound pressure of the reflected sound by -(0,42±0.14) times the sum of -(2,36±0.79), and if the value is negative, leave it as is. is positive, a third comparison conversion means outputs zero as the blur reference measure S, that is, outputs corresponding to the above-mentioned (Equations 20) to (Equations 22'); Relationship number 37
The value obtained by multiplying the square value by -(1,45±0.44) is output as the Brefference scale S4, that is, the above-mentioned (
A fourth comparing and converting means that outputs outputs corresponding to Equations 23) to (Equations 25) and preference scales S1. output from the first to fourth comparing and converting means. S2, S2, S
Since the sound field evaluation device is provided with a comprehensive evaluation means that comprehensively evaluates the sound field 4 and outputs the overall scale S of the sound field, it is possible to perform a faithful sound field evaluation.

(2) The acoustic device of the present invention: Next, the audible sound pressure in the sound field, the delay time of the first reflected sound, the reverberation time of the subsequent reverberant sound, and the interaural cross-correlation coefficient.
DESCRIPTION OF THE PREFERRED EMBODIMENTS An embodiment of the acoustic device of the present invention that can correct a sound field based on two physical parameters and create 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 an embodiment of the acoustic device of the present invention that can correct the sound field and create a more preferable sound field. 14 (BL is an example of a sound field expansion device included in the acoustic device of FIG. 14 (AL), FIG. 14 (C)
14(a) shows another example of the reverberation device included in the acoustic device of FIG. 14(a).

First, in FIG. 14(a), (501) is the input terminal I
NR1 (502) is the input terminal INL, (503a) ~
(503d) is an adder, (504a) (504b) is a reverberation device, (505a) (505b) (505r) (50
5f) is an attenuator, (506a) and (506b) are delay circuits, (507) is a scaler which is a type of frequency dividing circuit, and (5
08a) (508b) is an up/down counter, (5
09) is a digital-to-analog converter, (510a) (
510c) is a first reflected sound delay time analyzer, (510b)
are subsequent reverberation sound reverberation time analyzers, (511a) to (511
c) is an encoder, (512a) (512b) (513
a) (513b) is a comparator, (514a) (514b)
(514c) is a smoothing circuit, (515a) and (515b) are variable resistors for setting target values, (516) is a sound field expansion device, (517r) and (5171) are power amplifiers for driving speakers, (518r) (5184?
) is the speaker, (519) is the interaural cross-correlation coefficient (I
ACC) calculator, (520) is a multiplier, (531) (5
32) are input terminals inr, inl, (5
21) are CLOCK signal input terminals, (533) (534
) are the output terminals outr and ou of the sound field expansion device (516).
tf, (535) is an input terminal attm from the comparator (513b) to the sound field expansion device (516). 14th
In figure (b), (530a) to (530d) are adders, (536) is an attenuator, (537) is a low-pass filter (53,8) is a high-pass filter (5
39) is a phase shift circuit or a delay circuit. Figure 14 (C1
In, (504c) is M, R, Schroeder (S
Chroeder) and B, S.

Reverberator (54a) ~ (54) by Atal
0i) is an adder, (541a) (451b) is a phase inversion circuit, (542a) (542b) is a voltage follower, (543a) to (543f) are delay circuits, (544a)
) to (544f) are attenuators, (545) are attenuators (5
04c) input terminal, (546) is the attenuator (504C)
This is the output terminal of Note that the same reference numerals in the drawings as those in other drawings indicate the same or corresponding parts.

Next, the operation of an embodiment of the audio device of the present invention will be explained using FIG. 14(a) (bl (C1). First, record,
tape, microphone.

An acoustic signal from a radio or the like is given. These acoustic signals are added by an adder (503c) and input to the first reflected sound delay time analyzer (510c). This first reflected sound delay time analyzer (510c) is shown in FIG.
In other words, it is composed of the same circuit as that shown in Figure 13 (al), and in this case, the attenuation ratio is set to 1/lO so that the first reflected sound delay time can be measured. The output of the first reflected sound delay time analyzer (511) is input to the encoder (511).
), the delay time of the first reflected sound that is optimal for the sound source [
Output obtained by setting △t+]p to τp ([△t+)p=τ
p) appears. The output from this encoder (511c) is input to the comparator (512a), and the multiplier (511c) is input to the comparator (512a).
20) is also input, multiplied by 23, and output as the optimal subsequent reverberation sound reverberation time (Tsub) p (=23°τp) to the comparator (
512b).

On the other hand, microphones (2r) are installed at the left and right ears of a dummy head or human head (1) placed in a sound field such as a listening room (the signal coming in from 2I is used as a preamplifier C31). (3r), and the output is amplified by an adder (5r).
03d), and then inputted to the same first reflected sound delay time analyzer (510a) as shown in FIG. 12 as before, and also inputted to the subsequent reverberation sound reverberation time analyzer (510b). . This subsequent reverberation sound reverberation time analyzer (510b
) has the same circuit configuration as that described in the throat of FIG. Next, the first reflected sound delay time analyzer (5
The output of 10a) is input to the encoder (511a),
Delay time △t of the first reflected sound after being affected by the sound field
The time corresponding to 1 is obtained. After this, the comparator (512
a). In addition, the subsequent reverberation sound reverberation time analyzer (
The output of 510b) is input to the encoder (511b), and the subsequent reverberation sound reverberation time Ts after being affected by the sound field is
A signal corresponding to ub is obtained, and this signal is passed through a comparator (51
2b). By the way, in the comparator (512a), the optimum first reflected sound delay time [△t+) I) from the encoder (51IC) and the encoder (511
The delay time Δt1 of the first reflected sound after being affected by the sound field from a) is compared, and if the delay time Δt1 of the first reflected sound is greater than the optimal first reflected sound delay time [Δt,]p, When the delay time Δt1 is larger, the counter (508
A) is given a countdown signal CD, and the counter (5
08a), and for that purpose, the counter (5
A scaler (50
7), the value to be divided becomes small, and the input terminal (5
DEL obtained by frequency dividing the CLOCK signal from 21)
The frequency of the AY CLOCK signal increases, and its DEL
The acoustic circuit (504a) (5
Delay circuit (506a) configured by EIBD (packet brigade device) etc.
) (506b) becomes smaller. However, if the first reflected sound delay time Δt1 is smaller than the optimal first reflected sound delay time [Δt+)p, the counter (512a)
08a) is given a count-up signal CU, the contents of the counter (508a) become thick (and the scaler (507)
The value to be divided at becomes larger, so DELAY
The frequency of the CLOCK signal becomes lower and the delay circuit (50
6a) The delay time of (506, b) increases, which is the opposite of the above case.

Furthermore, in the comparator (512b), the multiplier (520
The optimum subsequent reverberation sound reverberation time [Tsub) p of ) is compared with the subsequent reverberation sound reverberation time Tsub after being given the influence of the sound field from the encoder (511b). Tsub) When the subsequent reverberation sound reverberation time Tsub is large with respect to p, the counter (5
A count-up signal CU is sent to the counter (508b), the contents of the counter (508b) are converted to an analog value by the digital-to-analog converter (509), and the smoothing circuit (
514c), then the reverberation circuit (504a) (504b)
are applied to the attenuators (505a) (505b), respectively, and the signals cause the attenuators (505a) (505b
) becomes large, and therefore the subsequent reverberant sound reverberation time Tsub is reduced.

However, if the optimal subsequent reverberation sound reverberation time (Tsub)
If the subsequent reverberation sound reverberation time Tsub is smaller than
is sent out, the contents of the counter (508b) are converted into analog values by the digital-to-analog converter (509), and then passed through the smoothing circuit (514C) to the attenuator (505a).
(505b), and the signal is applied to the attenuator (505b).
The attenuation value at 05a) (505b) becomes smaller, and therefore the reverberation time Tsub is increased, which is the opposite operation to the above case.

By the way, each amplified signal from the preamplifiers (3A) (3r) before being added is input to the binaural cross-correlation coefficient calculator (519), and the binaural cross-correlation coefficient IACC is calculated.
is determined and converted into analog, and the next smoothing circuit (514b
) is entered. Next, the comparator (513b) compares the signal from the smoothing circuit (514b) with the target voltage signal from the variable resistor (515b) that sets the target value of the interaural cross-correlation coefficient IACC. If the smoothing circuit (51
4b), that is, the signal corresponding to the binaural cross-correlation coefficient IACC, is larger than the target signal from the variable resistor (515b), the output signal from the comparator (513b) becomes smaller. , is input to the input terminal att 1n (535) of the sound field expansion device (516), and the sound field expansion device (516) operates to further reduce the binaural cross-correlation coefficient IACC.However, if The signal corresponding to the binaural cross-correlation coefficient IACC is
5b), the comparator (51
The output signal from 3b) becomes larger and is input to the input terminal att 1n (535) of the sound field expansion device, and the sound field expansion device (516) further increases the binaural cross-correlation coefficient LACC as described above. The operation is opposite to that of .

By the way, the first reflected sound delay time analyzer (510a), which receives the added signal added by the adder (503d), generates a sound pressure signal φ corresponding to the audible sound pressure. This sound pressure signal φ. After passing through the smoothing circuit (514a), it is input to the comparator (513a). In this comparator (513a), a variable resistor (5I5a) is used to set the target volume value.
) and the target voltage signal from the smoothing circuit (514a).
If the output signal from the smoothing circuit (514a) corresponding to the listening sound pressure is higher than the output signal from the variable resistor (514a),
If the volume from the comparator (513a) is larger than the target value, the output from the comparator (513a) increases and the attenuator (515a)
05r) (505j7) increases, thereby lowering the listening sound pressure. Moreover, if the signal corresponding to the listening sound pressure is smaller than the target volume value, the output from the comparator (513a) becomes smaller,
The attenuation of the attenuators (505r) (505β) is reduced to increase the listening sound pressure, which is the opposite of the above case.

Note that the smoothing circuit (514a) shown in FIG. 14(a)
) (514b) and (514C) are both designed to prevent the generation of noise due to sudden signal changes, and to use a reverberation circuit (514C).
Delay circuit (506a) (50
The delay time 6b) is also used to prevent sudden changes.

Next, the function of the sound field expansion device (516) is explained in Fig. 14(b).
) will be explained using the sound field enlarging device shown in FIG.

First, input terminal 1 from the reverberation device (504a) (504b)
nr (531) and 1njl! (532) entered 2
The difference components of the two signals are obtained by an adder (530a), passed through an attenuator (536), and then passed through a low-pass filter (53
7), the adder (530b) (530C) converts the input terminal 1nr (531) to the output terminal outr (53
3) and the input terminal inj! (532) to output terminal out 1. (534) Added to each main path up to 4. Apart from that, a low-pass filter (
The output of 537) is further passed through a high-pass filter (538
), a delay or phase shift rotation is applied in the phase circuit or delay circuit (539), and then an adder is added to each of the above-mentioned main paths with the same phase (, such that the crosstalking phases are opposite phases). (530d) and (530e), and the signal after the addition is the output terminal outr (533
) and outl C534), respectively.

By changing the attenuator (536) of this sound field expansion device (516), the interaural cross-correlation coefficient IACC can be changed. Further, this attenuator (536) is set so that the attenuation increases as the value of the signal from the input terminal att 1n (535), that is, the output signal from the comparator (513b) increases.

Furthermore, shown in Fig. 14(C) is a reverberator (504c) by M, R, Schroeder and B, S, Ata.
If this reverberation device (504c) is used in place of the reverberation devices (504a) and (504b) shown in FIG. 1, a more natural feeling of reverberation can be obtained.

In addition, in the above embodiment, the microphone (2r) (
2A) has been described to obtain the sound pressure at the entrance of the external auditory canal of a dummy head or a human head (1), but if it is not necessary to obtain the value of the interaural cross-correlation coefficient IACC, a single microphone may be used. However, even when calculating the interaural cross-correlation coefficient IACC, the
In some cases it may be possible to do so with a book microphone. Although the audio signal is shown as a two-channel stereo, it goes without saying that there are cases in which a two-channel signal is not necessarily required.

As described above, in the acoustic device of the present invention, the optimum first reflected sound delay time [△t
+) an optimal first reflected sound delay time measuring means for measuring p;
(23±1 of the optimal first reflected sound delay time [△t+)p
0) Optimal subsequent reverberation sound reverberation time (Tsub) corresponding to
l) optimal subsequent reverberation sound reverberation time output means for outputting;
The audible sound pressure φ in the sound field is determined by the sound field signal from the microphone placed in the sound field.

audible sound pressure measuring means for measuring the sound pressure, and a first
A first reflected sound delay time measuring means for measuring the reflected sound delay time Δt1, and a subsequent reverberation sound reverberation time Tsu based on the sound field signal.
b; subsequent reverberation sound reverberation time measuring means for measuring the interaural cross-correlation coefficient IACC from the sound field signal; and the above-mentioned optimal first reflected sound delay time [△ t+)p and the first reflected sound delay time Δt1, and outputs a signal according to the difference;
a second comparison means that compares the optimum subsequent reverberation sound reverberation time (Tsub) I) and the abovementioned subsequent reverberation sound reverberation time Tsub, and outputs a signal according to the difference; and an output signal of the first comparison means. a reverberation means whose delay time changes, whose reverberation time changes according to the output signal of the second comparison means, and which adds reverberant sound to the acoustic signal inputted from the input terminal; Compare the binaural cross-correlation coefficient setting means that can set the value of the relationship coefficient, the above-mentioned interaural cross-correlation coefficient IACC, and the set value of the above-mentioned binaural cross-correlation coefficient setting means, and calculate the difference between them. a third comparing means for outputting a signal according to the reverberation means; and receiving the output signal of the reverberation means, and changing the interaural cross-correlation coefficient IACC of the sound field according to the output signal of the third comparing means. a sound field expanding means having an output capable of increasing the audible sound pressure; a hearing sound pressure setting means capable of setting a target audible sound pressure value in advance; and an audible sound pressure φ of the sound field. and a fourth comparing means for comparing the set value of the audible sound pressure setting means and outputting a signal according to the difference, and receiving the output of the sound field enlarging means as an input and outputting the output of the fourth comparing means. Since it is equipped with attenuation means that can change the attenuation rate according to the signal, and electroacoustic conversion means that amplifies the signal from the attenuation means and radiates the acoustic signal into space, it corrects the sound field and makes it even more Not only can a favorable sound field be created, but also the optimal first reflected sound delay time [△t+)p and optimal reverberation time [Tsub]p can be obtained for the sound source within the acoustic device of the present invention, and the user can This has the effect that the listening sound pressure and the value of the interaural cross-correlation coefficient IACC can be changed according to the user's preference.

(2, + 2) Other acoustic devices related to the present invention Next, in a sound field device in a sound field such as a Japanese-style room with relatively few reflections or a car interior with many reflections (even if the reverberation time is extremely short), A sufficient and more preferable sound field can be created by correcting the sound field based on only three physical parameters: the audible sound pressure in the sound field, the first reflected sound delay time, and the subsequent reverberation sound reverberation time.

It goes without saying that the above-mentioned acoustic device of the present invention can be used as an acoustic device in the sound field targeted here, but the sound field used is in small rooms such as this, Japanese-style rooms with relatively low reflection, and cars. When limited to a room, other acoustic devices related to the present invention can not only obtain the same effect as the above-described acoustic device of the present invention without using the parameter of the interaural cross-correlation coefficient IACC. , the sound device is more compact and inexpensive, and the circuit configuration is simpler than the above-mentioned sound device, so it has better workability and is more durable than the above-mentioned sound device. can be played.

Therefore, this acoustic device is capable of creating a more preferable sound field by correcting the sound field using three physical parameters: audible sound pressure in the sound field, first reflected sound delay time, and subsequent reverberant sound reverberation time. Examples will be explained using figures.

FIG. 15 is a block diagram of an embodiment of another audio device related to the present invention.

In the figure, (451) to (453) are absolute value conversion circuits, (
511d, ) is the encoder, (551) is the input terminal IN
R1 (552) is an input terminal INt, and (553) is an integral reset signal input terminal. (554) is the first reflected sound delay time analyzer, and the circuit diagram of FIG. 13 (al) and the first reflected sound delay time analyzer (510a) of FIG.
) (510C), subsequent reverberation sound reverberation time analyzer (510b)
Although the internal configuration is slightly different from that of ), the functionality is almost the same, with an attenuator (42
6) and a comparator (419) are added. Even in the case of such a configuration, it operates in the same manner as the first reflected sound delay time analyzer (510a) (510C) or the subsequent reverberation sound reverberation time analyzer (510b) described above. In addition,
In the figure, the same reference numerals as those in FIG. 13(a) and FIG. 14(a) indicate the same or corresponding parts.

First, input terminals INII (551) and INL (55
2), the two signals are added by an adder (503C) and input to the first reflected sound delay time analyzer (544). This first reflected sound delay time analyzer (554) performs the same operation as the first reflected sound delay time analyzer (510a) (510c) in FIG. Figure 14 (
As in case a), the first reflected sound delay time [△t+
) p is output. This optimal first reflected sound delay time [△t
+]I) A signal from the encoder (511d) is given to the scaler (507) so as to obtain a delay time corresponding to I). The scaler (507) divides the frequency of the CLOCK signal from the input terminal (521) and outputs the DELAY CLOCK signal.
Create a signal and apply DELAY CLOC to the delay circuits (506a) (506b) of the reverberation circuits (504a) (504b).
A K signal is sent. The acoustic signal is input to the input terminal INR (551
) and INL (552) respectively to the reverberation circuit (504a
) and (504b), are given reverberant sound, and are sent to attenuators (505r) and (505j7). On the other hand, the first
The signal φ0 corresponding to the sound pressure of the reflected sound delay time analyzer (554) passes through the smoothing circuit (514a) and is sent to the comparator (513).
a) a variable resistor (
515a), and if the sound pressure is smaller than its target value, the output of the comparator (513a) becomes smaller, and the attenuator (505r) (505jl!) ), the amount of attenuation is reduced by the signal sent from the However, if the sound pressure is greater than its target value, the comparator (5
The output of 13a) increases, and the output of attenuator (505r) (5
05A) operates in the opposite manner to the above case, increasing the amount of attenuation by the signal sent from the comparator (513a). The output of the attenuator (505r) (505f) obtained in this way is the output of the power amplifier (505f).
17r) (via 517I, speaker (518r) (
518β).

At this time, the optimal subsequent reverberation time (
Tsub) I) is preferably (23±10) times the optimal first reflected sound delay time [Δt+)p, as described above. It is known that when a reverberator is configured in the form of a reverberation circuit (504a) (504b), the subsequent phonograph reverberation time is Tsub-1ε·Δt+/log(g). . Note that g is the attenuation rate. Here, by using the relationship Tsub=(23±10)Δt1, it can be seen that the attenuation rate g must be in the range of 0.588 to 0.811. In addition, when the subsequent reverberation sound reverberation time Tsub=23△t1, the attenuation rate g=0.7
It is 41.

That is, if the attenuation rate g is set in this way, the optimal first reflected sound reverberation time [△t+) p and the optimal subsequent reverberant sound reverberation time [T
sub3 p is obtained. Of course, even with such processing, effects in the sound field will be added, but in spaces with relatively little reflection, such as Japanese-style rooms, or spaces with a lot of reflection, such as car cabins, the reverberation time is extremely short. is very effective.

By the way, with this acoustic device, we aim to provide a simpler acoustic device that is as effective as or better than the aforementioned acoustic device in a small room or a sound field with relatively little reflection, such as a Japanese-style room or a car interior. However, when this acoustic device is used in a sound field other than a small room or a car interior, the performance is slightly lower than the above-mentioned acoustic device, but it still improves the audible sound pressure, first reflection delay time, and Since the sound field is corrected by three physical parameters: the subsequent reverberation time, it is possible to create a sound field that is much more favorable than that of conventional sound equipment; It has the advantages of being compact, inexpensive, has a simple circuit configuration, and is easy to work with and has good durability.

As described above, in this audio device, the input terminal IN
R, INL, an optimum first reflected sound delay time measuring means for measuring an optimum first reflected sound delay time from the acoustic signals inputted from the input terminals INR and INL, and a signal corresponding to the audible sound pressure from the acoustic signal. a signal measuring means corresponding to audible sound pressure, which receives a signal corresponding to the optimum first reflected sound delay time as a control signal, inputs the acoustic signal, and applies the acoustic signal to the optimum first reflected sound delay time. In addition to adding the corresponding delay time, the reverberation means is set so that the subsequent reverberation time is set to (23±lO) times the above-mentioned optimal first reflected sound delay time, and the value of the target listening sound pressure is set in advance. and comparing means that compares the audible sound pressure output from the audible sound pressure corresponding signal measuring means with a set value of the audible sound pressure setting means and outputs a signal corresponding to the difference. , attenuation means to which the output signal of the reverberation means is input, and which is capable of changing an attenuation rate given to the acoustic signal input from the reverberation means in accordance with the output signal of the comparison means; and a signal from the attenuation means. and an electroacoustic transducer that amplifies the acoustic signal and radiates the acoustic signal into the space.
In addition to correcting the sound field and creating a more favorable sound field, this acoustic device also provides the optimal first reflected sound delay time and optimal subsequent reverberation sound reverberation time, and adjusts the audible sound according to the user's preference. You can change the pressure.

As described above, according to the present invention, the acoustic characteristics of the sound field in which music is listened to are measured, and the audible sound pressure and first reflected sound delay time in the sound field are determined as physical and psychological quantities.

Based on evaluation using parameters such as subsequent reverberation sound reverberation time 1, interaural cross-correlation coefficient, and the like, it is possible to provide an acoustic device that changes the physical quantity by measuring this physical quantity and enables more preferable music listening. It has the effect of being able to do it.

In addition, comparators (512) (419) in the above-mentioned acoustic device
It is configured to obtain the output of (420) and create DELAY CLOCK using a scaler (507).
The first reflected sound delay time can be controlled by this DELAY CLOCK. This DELAY CLO
By making the CK value manually controllable from the outside, it is possible to configure it so that it can be changed in response to various types of music. Especially useful,
It has the advantage of being inexpensive to manufacture.

[Brief explanation of drawings]

Figure 1 is a diagram showing the relationship between a reflective wall, a sound source, and a human head, Figure 2 (al (b) is a diagram showing an example of a normalized autocorrelation function measured using actual music, and Figure 3 ( a) (b) is a diagram showing the results of a preference test conducted using music A and music B, Figure 4 is a diagram showing the relationship between red and [△t+)17, and Figure 5 is a diagram showing the relationship between τp and ( Tsub) Figure 6 shows the relationship between normalized preference and IA.
Figure 7 is a diagram showing the relationship between the values of CC, Figure 7 is a diagram showing the relationship between relative audible sound pressure and preference scale S1, and Figure 8 is a diagram showing the relationship between the direct sound and the first reflected sound and the preference scale. A diagram showing the relationship between the scale S2 and FIG. 9 is a diagram showing the relationship between the reverberation time of subsequent reverberant sound and the preference scale S3.
1I and 12 are block diagrams showing a schematic configuration of an embodiment of a sound field evaluation instrument related to the present invention, and FIG. Figures (a) and (b) are diagrams showing the basic configuration of a first reflected sound delay time analyzer or a subsequent reverberation sound reverberation time analyzer using a square integral type reverberation meter, and Figure 13 (c) (
d) is a diagram showing another example of the configuration of the audible sound pressure analyzer, FIG. 13(e) is a diagram showing an example of the configuration of a circuit for calculating the interaural cross-correlation coefficient, and FIG. 14(a) is a diagram showing another example of the configuration of the listening sound pressure analyzer. 14(b) and (C) are block diagrams of a sound field expansion device and a reverberator, and FIG. 15 is an implementation of another acoustic device related to this invention. FIG. 2 is a configuration block diagram of an example. In the figure, (1) is a human head or dummy head, (2r
) (2j') is a microphone, (3H3r) (3f
) is a preamplifier, (4) is a physical quantity analyzer, (5) is a comparator, (6) is a psychological quantity converter, (7) is a comprehensive evaluator, (
8) is the output terminal, (9) is the recorder, αO) is the sound field evaluation instrument, CD is the listening sound pressure analyzer, IAZr (510a)
(510c) (554) is a first reflected sound delay time analyzer;
U (510b) is a subsequent reverberation sound reverberation time analyzer, (Foundation) is a binaural cross-correlation function analyzer, Sl) is an audible sound pressure psychological quantity converter, ■ is a first reflected sound delay time psychological quantity converter, ■ is a subsequent reverberation sound reverberation time psychological quantity converter, the figure is a binaural cross-correlation coefficient psychological quantity converter, (451) to (453) are absolute value conversion circuits, (501) (502) (521) (531) (532
) (535) (551) (552) are input terminals, (50
3a) to (503d) are adders, (504a) (504
b) is a reverberation device, (505a) (505b) (505r
) (505β) is an attenuator, (506a) (506b) is a delay circuit, (507) is a scaler, (508a) (5
08b) is an up/down counter, (509) is a digital-to-analog converter, (511a) to (511d) are encoders, (512a) (512b) (513b) (
513b) is a comparator, (514a) to (514C) are smoothing circuits, (515a) and (515b) are variable resistors, (5
16) is a sound field expansion device, (517r) (51M) is a power amplifier, (518r) (518f) is a speaker,
(519) is a binaural cross-correlation coefficient calculator, (520)
is a multiplier, and (533) and (534) are output terminals. In addition, in the figures, the same reference numerals indicate the same or corresponding parts. Figure 1

Claims (1)

[Claims] (1) an input terminal, an optimum first reflected sound delay time measuring means for measuring an optimum first reflected sound delay time from an acoustic signal input from the input terminal; 10) Optimum subsequent reverberation sound reverberation time output means that outputs the optimum subsequent reverberation sound reverberation time corresponding to 100%, and listening sound pressure that measures the listening sound pressure in the sound field from the sound field signal from the microphone placed in the sound field. a measuring means, a first reflected sound delay time measuring means for measuring a first reflected sound delay time from the sound field signal, a subsequent reverberation sound reverberation time measuring means for measuring a subsequent reverberation sound reverberation time from the sound field signal; A binaural cross-correlation coefficient measuring means for measuring a binaural cross-correlation coefficient from the sound field signal compares the optimum first reflected sound delay time and the first reflected sound delay time, and calculates the difference between the binaural cross-correlation coefficients. a first comparison means for outputting a signal according to the difference; a second comparison means for comparing the optimum subsequent reverberation sound reverberation time and the abovementioned subsequent reverberation sound reverberation time and outputting a signal according to the difference; Reverberation means for adding reverberation sound to the acoustic signal inputted from the input terminal, wherein the delay time changes according to the output signal of the first comparison means, the subsequent reverberation sound reverberation time changes according to the output signal of the second comparison means, and the reverberation sound is added to the acoustic signal inputted from the input terminal. , a binaural cross-correlation coefficient setting means for setting a target value of the binaural cross-correlation coefficient in advance; and setting of the binaural cross-correlation coefficient and the binaural cross-correlation coefficient setting means. a third comparing means for comparing the signals and outputting a signal corresponding to the difference; and a third comparing means for receiving the output signal of the reverberation means;
sound field expansion means having an output capable of changing the interaural cross-correlation coefficient of the sound field according to the output signal of the comparison means; and a listening sound pressure capable of setting a target listening sound pressure value in advance. a setting means, a fourth comparison means for comparing the audible sound pressure of the sound field and a setting value of the audible sound pressure setting means, and outputting a signal corresponding to the difference; Attenuation means that receives as input and can change the attenuation rate according to the output signal of the fourth comparison means, and electroacoustic conversion means that amplifies the signal from the attenuation means and radiates the acoustic signal into space. A sound device characterized by comprising: