JPS6162824A - Method and apparatus for compressive intake of reverberation data - Google Patents

Abstract

PURPOSE:To reduce memory capacity and to simplify the constitution of a post-processing circuit, by converting sound collection input to digital data which is, in turn, squared, cumulated and added while receiving the cumulated and added data in memory. CONSTITUTION:The first impulse from a sound source 3 is collected by a microphone 4-1 and the collected impulse response is amplified by an amplifying circuit 6 to be supplied to an analogue/digital converter circuit 7. The signal converted by the analogue/digital converter circuit 7 is squared by a square circuit 8 and inputted to an accumulator 9 to be successively cumulated in the timing of a signal C-1. This cumulated value is successively inputted to a register 10 in the timing of a signal C-3 and the content of the register 10 is successively added to the content of a register 12 in the timing of the signal C-3 by an adder circuit 11. This added result is stored in a random access memory 14 and inputted to a register 13.

Description

DETAILED DESCRIPTION OF THE INVENTION [Field of Industrial Application] The present invention relates to a reverberation data compression/intake method and apparatus used for measuring reverberation characteristics of an indoor space.

[Prior art]

For example, when a lecture is held in a hall or a musical performance is performed by an orchestra, if the reverberation characteristics of the square hall are accurately understood, it is possible to aim for a high acoustic effect during the lecture.

These reverberation characteristics include the following.

■ Reverberation time (n, Tso value) The time required for the sound energy at the sound receiving point to attenuate by 60 dB after the sound source stops.

Indicates the level of reverberation of sound in the room.

■ Initial reverberation time (EDT + o value) After the sound source stops, the sound energy at the sound receiving point is 10 dB.
Time required for decay.

It is generally said that the feeling of reverberation is more closely related to this initial reverberation time EDT+o than to the reverberation time 11 T 6o.

■D value The value given by the following equation is called the D value.

0 indicates the degree of reverberation of the sound and the clarity of the voice at the indoor listening point. ■ Reverberation waveform The state of attenuation of reverberant sound is displayed on a CRT or the like.

Furthermore, the following methods are known as methods for measuring these reverberation characteristics.

(b) A method of cutting off continuously occurring band noise (noise sound extracted from white noise through a bandpass filter) and measuring the attenuation characteristics of the sound receiving point at this time.

In principle, measurements made using the above method will yield different results for each measurement, so in order to obtain accurate reverberation characteristics, it is necessary to repeat the measurements many times and take the average.

(b) Impulse response square integral method.

This method was proposed by M. R. 5Chroeder, and it calculates the characteristic corresponding to the average of (1) measurement of the attenuation characteristic of the sound receiving point when band noise is cut off, between the sound source and the sound receiving point. Impulse response r (X) K M is determined by one measurement. In this method, the set average <s2(t)> of infinite times of the square of the sound pressure level S (t) (corresponding to the energy value) at time t when the sound is in a decay state is expressed as <82(t) >-
N f r 2(x) d x −,・
,, (21, where N: variation of sound source band noise, r(X): impulse response. In other words, this impulse response square integration method calculates the reverberation characteristics based on the above equation (2). It is a method of measurement.

[Problem that the invention seeks to solve]

Now, when measuring the reverberation characteristics by the method (a) or (b) mentioned above, the sound pressure level is usually lower than 5 at the sound receiving point.
(t) or its square 52(t) is sequentially sampled and stored in a memory, and the data in this memory is read out to measure characteristics. Particularly in the case of method (b), characteristics cannot be measured unless - is stored in the memory. Conventionally, the output signal of a microphone installed at a sound receiving point is amplified and then sequentially sampled.
D (analog/digital) conversion is performed, and all the digital data thus obtained is stored in a memory, and then characteristic measurement calculations are performed based on the data in this memory. However, in the above-mentioned method of capturing reverberation data into memory, a large amount of memory is required (particularly when the sampling clock is set to a high speed to improve accuracy, or when the reverberation time of the system itself is long). There is a problem that the memory capacity becomes enormous, and in the worst case, the necessary data may be recorded only halfway.

The present invention has been made in view of the above circumstances, and its object is to provide a method and apparatus for compressing and capturing reverberation data, which can reduce memory capacity to a much smaller extent than in the past.

[Means to solve the problem]

This invention samples a sound input in accordance with a basic clock, converts this sampling result into digital data, squares this digital data, and cumulatively adds the squared results to obtain cumulative addition data. The present invention is characterized in that the cumulative addition data is sequentially loaded into a memory in accordance with a low-speed clock of a lower frequency, and processing for measuring reverberation characteristics is performed based on the cumulative addition data loaded into the memory.

[For production]

This invention does not directly store digital data obtained by A/D conversion of sound input into memory,
Since the digital data is squared and then cumulatively added, and the cumulatively added data is stored in the memory, the memory capacity can be much reduced compared to the conventional method.

〔Example〕

FIG. 1 is a block diagram showing the configuration of a first embodiment of the present invention. The embodiment shown in this figure is an apparatus for measuring reverberation characteristics using the above-mentioned impulse response square integration method. Further, in this embodiment, it is possible to obtain a comprehensive characteristic (spatial collective average) of the entire room, which is a synthesis of the reverberation characteristics at n sound receiving points in the room.

In Fig. 1, the output of a short sound generator 1 that generates impulses (filtered impulses) for each frequency band is output from a sound source 3, for example, a speaker, installed in a room 2 in which the reverberation waveform is to be measured. is input. A large number of microphones 41 (1=112...n) are installed in the same room 2, and this Keikuro7on 41 (i=L2...
・n) is each contact 51 of the changeover switch 5 (1=11
．． 2...n), and the common terminal 5-C0M of this selector switch 5 is connected to the input terminal of an amplifier circuit 60, and the output of this amplifier circuit 6 is connected to an A/D converter circuit 7, a squaring circuit ( It is input to the accumulator 9 via the multiplier 8. This accumulator 9 has an addition register inside,
It has a built-in full adder, etc., and its output is in register 10.
The signal is inputted to the adder circuit 11 via the adder circuit 11, and is also inputted to the other input terminal of the accumulator 9. The addition circuit 11
is for adding the output of register 1° and the output of register 12, and the output is input to register 13 and also to random access memory (hereinafter referred to as RAM).
) 14, and the output of this RAM 140 is input to the register 12 as well as to the subtraction input terminal Ta of the subtraction circuit 15, and the output of the register 13 is passed through the latch circuit 16 to the subtraction circuit 15. subtracted input terminal T
bK is input. The output of the subtraction circuit 15 is input to a display/recording device 19 via a read-only memory (hereinafter referred to as ROM) 17 and an interface circuit 18, where it is displayed/recorded.

In this figure, the broken lines indicate control signals from the timing control circuit 20, and the signals C-1, 1,
The signal C-2 is input to the register 12, and the signal C-3 is input to the register 1.
0, the register 13, the RAM 14, and the latch circuit 16, and the signal C-4 is input to the RAM 14.

Next, the operation of the above embodiment will be explained with reference to the time chart shown in FIG. First, the first impulse is sound source 3.
The common terminal 5-C0 of the changeover switch 5
Connect M and contact 5-1, and also connect accumulator 9 and R.
After clearing the AM 14 and the register 12, the first impulse from the sound source 3 is collected by the microphone 4-1. This collected impulse response rt (
x) is amplified by the amplifier circuit 6 and then converted to the A/D converter circuit 7.
supplied to A signal C-1 is input to the A/D conversion circuit 7 from the timing control circuit 20.

This signal c-1 is generated by the A/D as shown in FIG.
This is a basic clock pulse used during sampling and A/D conversion operations in the conversion circuit, and samples the analog signal input to the A/D conversion circuit 7, and converts this sampled value into the A/D conversion circuit 7. Convert to digital signal. This A/D converted digital signal is squared (multiplied) by a squaring circuit 8, and this squared value is input to an accumulator 9.
sequentially accumulates the input digital signals at the timing of the signal C-1.

This accumulated value is determined by the timing of the signal C-3 (low-speed clock) from the timing control circuit 20 (see Fig. 2 (b)).
), and the contents of this register 10 are input to register 10 at the timing of signal C-3.
2 and the adder circuit 11, and the result of this addition is stored in a predetermined location of the RAM 14 and input to the register 13. It is assumed that the RAM 14 has an address structure as shown in FIG.
). Further, as is clear from FIG. 2(b), the signal C-3 is a clock pulse having a lower frequency than the signal C-1 (basic clock).

Now, from the measurement start time (t=0), the contents of the accumulator 9 (supposed to be Sr-1) are changed by the first pulse of the signal C-3 (the pulse at time 1-1 shown in FIG. 2 (b)). When the data is input to the register 10, at the same time, the contents of address 1 of the RAM 14 are input to the register 12 by the signal C-2 (shown in FIG. 2) of the timing control circuit 20 (the RAM 14 is initially cleared). Therefore, the contents of address 1 are '0'').The contents of the input register 12 (D'') and the contents of the register 10 (Sl-+) are added by the adder circuit 11 (Sl-++0), and the contents are stored in the RAM. ]
It is written again to address 1 of 4 and is also input to register 13. Next, at time t1-2 shown in FIG. The addition circuit 11 adds (Sl-2+O), and this addition result (Sl-2) is written to address 2 of the RAM 14 again and is input to the register 13.

In the same manner, the RAM 14 is stored until the accumulation end point t1-m.
Writing and input to the register 13 are performed sequentially.

That is, when writing at time tl-m is completed, R
Each address of AM14 has address 1→S 1-12→
Address 5l-2,3 → Address 5l-3,... m → 51-m
Each data is transferred to register 13 again! St-
m has been input.

In this way, when the data collection of the impulse response rl(X) in the microphone 4-1 is completed,
After the common terminal 5-C0M of the changeover switch 5 and the contact 5-2 are connected and the accumulator 9 is cleared, the second impulse from the sound source 3 is picked up by the microphone 4-2.

The sonoimpulse response r2(X) is sequentially accumulated in the accumulator 9 via the amplifier circuit 6, A/D conversion circuit 7, and squaring circuit 8, as described above.

Now, at time t2-t (see Fig. 2), the first accumulation jF(K (52-
1) is input to register 10, at the same time RA
The contents of address 1 of M14 (51-1 is stored in the above process) are input to the register 12, and the adder circuit 1
1, the calculation of (52-1+5I-1) is performed, and the result of this calculation (52-1+5I-1) is 1 of 11AM14.
The address is inputted again and also inputted to the register 13. Similarly, impulse response r2(
The accumulation of X) 1) Hfk is sequentially added to the contents of the RAM 14, and at the end of the accumulation time t2-m, each address of the RAM 14 has address 1→[81-1+82-1.11. 2
Address → (31-282-2.11゜3-+ (S
1-3 + S 2-a]--” address → [5l-t
Each data of II+S2-m) is written, and the value of (St-m+s+-m) is input to the register 13.

Next, microphone 4-3.4-4.・・・・・・
4- Inva A/X Lesbore slot r 3 (x
), r4(x)...rn(x) are sequentially collected, and the accumulated values are sequentially added to the RAM 14. That is, n impulse responses rt (x),
r2 (x)...rzl(→ When all the processing is completed, the contents of the RAM 14 are as follows:
S+-tl+82-1+...+Srut=TSl).

Address 2-+ (St-2+82-2 +...+S n
-z = T 82:), ...”'m address → CSt
−m+sz−m+·”+sn−m=Ts m ], and the value of this TSm is input to the register 13.

Therefore, the contents mm of the register 13 are the same as those of the latch circuit 16.
The answer in the RAM 140 is sequentially read from address 1 and supplied to the subtraction input terminal Ta of the subtraction circuit 15.
5, the following calculations are performed in sequence. That is, [TSm-TS+=Zl], (TSm-TSz=Z
z) Hit・(TSm'PTSm=Zm:] , is given.The results of each calculation Z1-Zm are stored in ROM1
The data is logarithmically compressed using the logarithmic operation (1010g In addition,
The output waveforms of each part of the above device are shown in FIG. In this figure, (a) is the impulse response rt(X) (1=1.2...n) input to the amplifier circuit 6.
, (b) shows the output of the square circuit 8, (c) shows the output of the register O in analog display for convenience, and (b) shows the n impulse responses r1 (
For convenience, the contents stored in the RAM 14 at the time when all measurements of X) (1=1, 2, river...n) are completed are shown in an analog display, and (E) is the timing generator's timing generator. The signal C-3 is shown again, (f) shows the output of the subtraction circuit 15 in an analog display for convenience, and (g) shows an example of display/recording in the display/recording device 19. It shows.

Therefore, each of the above cumulative values (81-1 to Sn-m.

When T S1 to T S In, zl to Z m ) are expressed as integral expressions, they are as follows. That is, S1-j=ftjr1''(x)dx. Therefore, the above-mentioned device achieves spatial averaging of the reverberation waveform by the following relational expression.

Now, in addition to the above-mentioned functions, the above-mentioned apparatus has a function of displaying the intermediate progress of each measurement, so that the measurement can be performed while referring to the intermediate progress. Next, this function will be explained. For example, the signal C-5 of the timing control circuit 20 causes the RAM1
Assume that a new cumulative value is written to address g of No. 4, and this cumulative value is input to the register 13. Next, the RAM address signal C-4 of the timing control circuit 20 is output as shown in FIG.
As is clear from c), the address is changed from address 1 to address m until the next pulse of the signal C-3 arrives, and the contents of the RAM 14 are sequentially changed to the subtraction input terminal T of the subtraction circuit 15.
supply to a. On the other hand, the accumulated value input to the register 13 is input to the subtracted input terminal of the subtraction circuit 15 via the latch circuit 16, and in the subtraction circuit 15, [(content of register 13) - (content of RAM address 1)]
[(Contents of register 13 →-(RAM2-Contents of occupied land)
] [(Contents of register 13) - (Contents of RAM address 1)] are sequentially subtracted, and the result of this subtraction, that is, the reverberation waveform at that point in time, is sequentially displayed on display device 19.

That is, the above-mentioned arithmetic processing is performed every cycle of the pulse of the signal C-3, and the progress of the final reverberant waveform of the measurement H is displayed. Note that FIG. 5 shows an example of the display of this platform.

In this way, the above device achieves spatial collective averaging of reverberant waveforms using a large number of microphones;
As shown in Fig. 1, by sequentially moving one microphone, the impulse response r1(X
) can also be used to obtain a similar effect. Further, the above-mentioned device is used in a reverberation chamber having rotary blades.
When averaging the reverberant waveform at a certain point (the curve is different each time depending on the state of the rotor), when obtaining a spatial ensemble average of the reverberant waveform by changing the sound source position instead of the sound receiving position, or when obtaining a spatial collective average of the reverberant waveform by changing the sound source position instead of the sound receiving position, It can be used in an extended manner to obtain on-axis averaging (for example, averaging of waveforms of 100'Hz and 20011 Hz) or to obtain composite characteristics of each of the above cases.

Next, a second embodiment of the invention will be described. Third
The figure is a block diagram showing the configuration of a second embodiment of the present invention, and the embodiment shown in this figure includes the method (a) described above,
In other words, it is a device that measures reverberation characteristics based on band noise. Furthermore, this embodiment is also designed to measure the overall characteristics (spatial collective average) of the entire room.

The difference between the embodiment shown in this figure and that shown in FIG.
A band noise generator 21 is provided instead of the short tone generator 10 in FIG. 1, and the register 1 in FIG.
3. A delay circuit 22 is provided instead of the latch circuit 16.

This delay circuit 22 is constituted by, for example, the same number of delay flip-flops as the number of bits of the output data of J(AMI 4), and outputs the output data of the RAM 14 after being delayed by one change timing of the address signal C-4.

In addition, in FIG. 3, the subtraction circuit 150 input terminal Ta,
Tb is shown in the opposite position to that in FIG.

Next, the operation of the above embodiment will be explained.

First, common terminal 5-C0M of changeover switch 5 and contact 5-
1, the speaker 3 continuously generates band noise, and then the band noise is turned off. At the same time as this disconnection, accumulator 9, registers 10 and 12
, RAM 14, delay circuit 22, and then
A start signal is output to the M/G control circuit 20. Thereafter, a signal C similar to that shown in FIG. 2 described above is sent from the timing control circuit 20.
-1 to C-4 are output, and in the above case and inversion, RA
Data sl-+1Sl- at addresses M1401 to M, respectively.
2*..., Sl-m (see FIG. 7(c)) is written.

In the embodiment shown in FIG. 3, the sound generated from the speaker 3 is not an impulse but a band noise.
Therefore, the input signal to the amplifier circuit 60 is the reverberation signal itself, and in other words, the above-mentioned sound pressure level S (t
). FIG. 7(r) shows the input signal of this amplifier circuit 6. As a result, the output of the square circuit 8 (FIG. 7 (b)) becomes the square of the sound pressure level S (t), S''(t), which is a value corresponding to the energy of the reverberant sound. The data 5t-t written in is the time t when the band noise was cut off.

(See Figure 7) to time tl-1 (See Figure 7 (E))
This value is the cumulative value of the square 52(t) of the sound pressure level S (t) sampled up to this point. Similarly, RAMI 4
The data 51-2 stored at address 2 is at time 1 (,~t
The value obtained by accumulating the square 52(t) of the sound pressure level 5(t) sampled between 1 and 2,..., R
Data S 1 + m in address m of AM14 is at time t
. -tl It is the value obtained by accumulating each data S''(t) between -m.

As a result, data 5l-s to 51-m are all I(, A
, M14, if the data in the RAM 14 is sequentially read from address 1 and supplied to the subtraction circuit 15 and the delay circuit 22, the subtraction circuit 15 first reads the data 81-
1 is output (at this time, the output of the delay circuit 22 is 0 due to the initial reset), and then data (81-
281-1) is output, and below, the data (Sl-s −
81-2L (Sl-45l-3)+・・・・・・(
Sl-m-81(Iff-+)) are sequentially output.

Then, each output data is supplied to the display/recording device 19 via the ROM 17 and the interface circuit 18, and is displayed/recorded. Here, data 51-1 is at time t
Accumulated value of 52(t) during o''tl-1, data (5l-
25t-1) is S''(t;
) cumulative value,..., data (Sl-51-m-
5s-(>) is the cumulative value of 52(t) between time tl-(m-1) and t, -m. Also, the time to""'tt
-tt1-1 to tl-21... each time interval is equal (the number of sample points is equal). As a result, each output data of the subtraction circuit 15 is 82(t
), and therefore, the display/recording device 19 sequentially displays/records the average value of the reverberant sound energy at the installation point of the microphone 4-1.

Note that the address signal C-4 changes from "1" to "m" during one cycle of the signal C-3 as shown in FIG. m
The above display will be performed during one cycle of the signal C-3 immediately after.

Next, the common terminal 5-C0M of the changeover switch 5 and the contact 5-
2 is connected, band noise is generated from the speaker 3, and then the band noise is cut off, and the accumulator 9, the register 10° 12, the delay circuit 22 are cleared, and the start signal is output to the timing control circuit 20. When this is done, all data is written into the RAM 14 in the same process as above. However, in this case, the data collected by the microphone 4-1 has been written in the RAM 14 in advance, so The sum of each piece of data collected by the microphones 4-1 and 4-2 is written in. As a result, the display/distribution device 19 immediately after the 1. entry of address m of RAMI 4 is completed.
, the sum of reverberant sound energy at the positions of microphones 4-1 and 4-2 is displayed.

Thereafter, when data recording is performed sequentially using microphones 4-6 to 4-n in the same manner, when all data recording is completed, data T is stored in addresses 1 to m of RA and M14 respectively.
S 1 ~ T S m ('1.'5k=S1-10S
+...+Sη-k) is written to 2-. Note that FIG. 7) shows data TS, to TSm, and FIG. 7(e) shows signal C-3. Next, based on the data T S s - T S m written in the RAM 14, display is performed on the display/recording device 19 in the same manner as in the case described above. As a result, microphones 4-1~
The sum of the reverberant sound energies of each 11"L ft of 4-n is displayed. FIG. It is a figure which shows the display process in .

Next, a third embodiment of the invention will be described. 9th
The figure is a block diagram showing the configuration of a third embodiment of the present invention, and the functions of this embodiment as an apparatus are the same as those of the second embodiment shown in FIG. The difference in structure between this third embodiment and the second embodiment is that the subtraction circuit 15 and delay circuit 22 in FIG. 3 are not provided in the one in FIG. 9, and the accumulator in FIG. 9 is signal C
The point is that it is reset by -3.

In this embodiment, the time t when the band noise is cut off is
The square 52(t) of data S (t) sampled from o to time 11-1 is accumulated by the accumulator 9, and this accumulation result is written to the RAM address 1401 at tl-1. At the same time, the accumulator 9 is reset, and then at time t1
Data S sampled between -1 and tl-2
(t) squared 52(t) is accumulated by the accumulator 9, and this accumulation result is R at time t1-2.
It is written to address 2 of AM14, and the same process is repeated. That is, according to this embodiment, signal C-3
The accumulated value of 82(t) for each cycle of is sequentially written into the RAM 14, and therefore, each data in the RAM 14 is sequentially written to the ROMI 7 and the interface circuit 18.
If the data is outputted to the display/recording device 19, a display similar to that shown in FIG. 8 will be displayed.

In addition, the above-mentioned 2. Since both of the third embodiments perform measurements based on the reverberant signal itself, as mentioned above,
It is not possible to accurately measure reverberation characteristics with only one measurement, but it is necessary to repeat the measurement many times. In this case, the above 2. In the third embodiment, unless the RAM 14 is reset, the measurement results are accumulated in the RAM 14 one after another, and therefore, the measurement results are automatically averaged without performing any averaging processing of the measurement results of a large number of times. You can get results.

〔Effect of the invention〕

As explained above, according to the present invention, sound input is converted into digital data, this digital data is squared, cumulatively added, and this cumulatively added data is recorded in memory, thereby reducing memory capacity. In addition, the configuration of the post-processing circuit based on the data in the memory can be simplified. Furthermore, when collecting sound input from multiple locations in the measurement space, a spatial collective average of reverberation characteristics can be obtained.

[Brief explanation of drawings]

FIG. 1 is a block diagram showing the configuration of a first embodiment of the present invention, FIG. 2 is a timing diagram of signals C-1 to C-4 in the same embodiment, and FIG. 3 is an address of the RAM 14 in the same embodiment. FIG. 4 is a diagram showing the waveform of each part of the same embodiment. FIG. 5 is a diagram showing the display state in the display/recording device 19. FIG. 3 is the configuration of the second embodiment of the present invention. FIG. 7 is a waveform diagram showing the waveforms of each part of the same embodiment. FIG. 8 is a diagram showing the display state of the display/recording device 19 in FIG. 3. FIG. FIG. 2 is a block diagram showing the configuration of an embodiment of the present invention. 4-1~4-n...Microphone, 7...
...A/D conversion circuit, 8... Square circuit, 9
...Accumulator, 11...Addition circuit, 14...RAM, 20...
Timing control circuit, C-1... signal (basic clock), C-6... signal (low speed clock). (c) Fig. 5 Fig. 4 (/l) (c) Fig. 8

Claims (4)

[Claims] (1) Sampling the sound input according to the basic clock, converting this sampling result to digital data, squaring this digital data, cumulatively adding the squared result to obtain cumulative sum data, and The cumulative addition data is sequentially taken into memory according to a low-speed clock having a frequency lower than that of the clock, and processing for measuring reverberation characteristics is performed based on the cumulative addition data taken into the memory. Reverberation data compression and import method. (2) The sound input is obtained from a plurality of locations in the measurement space, and the operation based on the basic clock is sequentially and repeatedly performed for each of the sound input from each of these locations, and the Each loading operation into the memory according to a low-speed clock is performed by additive multiple loading into a corresponding storage area of the memory, and a spatial collective average of reverberation characteristics is measured based on the loaded data. A method for compressing and capturing reverberation data according to claim 1. (3) sound collection means, analog-to-digital conversion means for sampling the output of this sound collection means and converting it into digital data, squaring means for squaring the digital data from this conversion circuit, and output data of this squaring means. a cumulative addition output stage for cumulatively adding the data, a storage means for storing the data of the cumulative addition means, a calculation means for calculating the reverberation characteristics by calculating the data stored in the storage means, and timing for generating at least high-speed and low-speed clocks. control means, the analog-to-digital conversion means, the squaring means, and the cumulative addition means are given operating timing by a high-speed clock generated from the timing control means, and the storage means is provided with an operation timing generated by the high-speed clock generated from the timing control means. 1. A reverberation data compression/capturing device, characterized in that the reverberation data compression/capturing device is configured such that the timing of data fetching operation is given by a low-speed clock. (4) A plurality of the sound collecting means are provided in the measurement space, and a switching means is provided for sequentially switching the sound collecting means so as to obtain the cumulative addition data for each output of each sound collecting means, and a memory; The memory is equipped with an address control means for controlling the address of the storage memory so as to add and overwrite the accumulated data of each of the sound collection means, and is configured to measure the spatial collective average of the reverberation characteristics. A reverberation data compression/intake device according to claim 3, characterized in that: