JPS61261997A - Controller for collecting and reproducing sound - Google Patents

Abstract

PURPOSE:To make t possible to feel realistic atmosphere where a player himself/ herself is playing at a hall, etc. by simulating reflected sound in a performance based on the data of the reflected sound at an actual hall and the like. CONSTITUTION:Reflected sound data 40 (a direction, delay time and an amplitude level) from a virtual sound source at the hall and the like are obtained from the distribution etc. of the virtual sound source at the hall and the like. And at a control factor 42, a reflected sound parameter to generate the reflected sound to be vocalized from each of speakers 56-62 to reproduce the hall and the like with the reflected sound data 40 and the arrangement of the speaker is obtained. And at a processor 46, the reflected sound of a microphone collecting sound signal which is a source 44 based on the reflected sound parameter is generated and is vocalized from the speaker 56-62.

Description

[Detailed Description of the Invention] [Industrial Field of Application] This invention is applicable to J3 in a general room or test chamber,
The present invention relates to a sound collection and playback control device that can create an atmosphere as if a performance (playing an instrument, singing, etc.) is being performed in a large hall or the like.

[Conventional technology]

When playing instruments such as the piano or flute or singing in a regular room, it would be much more enjoyable if the performers could feel as if they were playing in a large hall. It can be done. Also, when professional musicians practice in a test room, etc., it creates an atmosphere similar to being in an actual hall (
If you can perform with the correct tone (including tone, etc.), you will be able to practice more effectively and in line with the actual performance.

Conventionally, reverberators, karaoke systems, surround processors, and the like have been proposed as devices that add a hall-like feel and a sense of spaciousness. As shown in FIG. 2, these signals are processed by a processor 14 such as adding appropriate reverberation to the original sound using the source 12 itself (musical instrument performance sound or singing sound) as a control factor 10, and an amplifier 16.
18 to the loudspeakers 20.22.

However, these are simply added artificial reverberations, which are completely different from the natural reflected sounds and reverberations in an acoustic space such as a hall, and cannot create the atmosphere of a performance in an actual hall. Ta.

[Problem that the invention seeks to solve]

The present invention aims to solve the problems in the conventional techniques and provide a sound collection and playback control device that can create the atmosphere of a performance in an actual hall or the like in a normal room, test chamber, etc. It is something.

[Means for solving problems]

This invention places a plurality of speakers around a performance position in a normal room, etc., and simulates the reflected sound in that acoustic space or a similar model space based on reflected sound data in the acoustic space such as a hall. As if to
The parameters of the reflected sound to be emitted by each speaker (reflected sound parameters) are determined, the performance is picked up by a microphone, and the reflected sound is generated based on the reflected sound parameters.

Note that the "reflected sound data" here refers to the data that constitutes the reflected sound in the acoustic space, and specifically, the direction and distance (= delay time) of the reflected sound determined from the virtual sound source distribution, etc. and data such as amplitude level.

In addition, the "reflected sound parameter" is used to generate the reflected sound that should be emitted from each speaker in order to simulate the reflected sound specified by the anti-OJ g data using multiple speakers placed around the performance position in the room. Specifically, they are parameters for delay time and gain. This reflected sound parameter is determined based on the relationship between the reflected sound data and the position of the speaker relative to the performance position.

[Effect]

According to the solving means of the present invention, the reflected sound of a performance is simulated based on the reflected sound data of an actual hall, etc., so that the performer can enjoy the atmosphere of playing in the hall.

[Principle of the invention]

In this invention, the direction, delay time, amplitude level, etc. of reflected sound are used as reflected sound data. These reflected sound data can be obtained, for example, from a virtual sound source distribution in an acoustic space such as a hall. Here, the virtual sound source refers to an effective sound source of reflected sound viewed from a specific sound receiving point in an acoustic space such as a hall. In other words, the sound emitted from the actual sound source (referring to the actual sound source) not only reaches the sound receiving point as direct sound, but also reflects from all reflective parts in the acoustic space such as walls, ceilings, floors, seats, etc. and reaches the sound receiving point. In this case, at the sound receiving point, the reflected sound can be regarded as the sound emitted from the sound source that is on the extension of the line connecting the sound receiving point and the reflection point, such as a wall surface. It can be understood as a virtual sound source.

Therefore, the acoustic space at a certain sound receiving point can be understood as the distribution of virtual sound sources at that sound receiving point, and even in a normal room, if the reflected sound of the performance from each virtual sound source is simulated, It is possible to reproduce that acoustic space and enjoy the atmosphere of actually playing in that acoustic space.

The position of a virtual sound source is determined by the direction and distance from the sound receiving point, so in order to simulate the reflected sound from the virtual sound source, the direct sound picked up by the microphone is determined by the direction and distance from the virtual sound source. It is sufficient to emit the sound with a time delay corresponding to the reflected sound and at a volume corresponding to the amplitude level of the reflected sound. If this is done for each virtual sound source in the acoustic space, it is possible to reproduce the state of playing in that It space.

There are two ways to find a virtual sound source: one is to actually measure the impulse response in an acoustic space such as a hall to be reproduced, and the other is to calculate it from the shape of the acoustic space such as a hall.

■Method to obtain by measurement As the former method by measurement, there is a so-called four-point method. This uses the time difference between the impulse responses of four adjacent points in the N-sound space to determine the coordinates of a virtual sound source as seen from those points.

Impulse response is considered to be the sound signal picked up at the sound receiving point when impulses are emitted simultaneously from a real sound source and a virtual sound source.In the initial part of the response, reflected sounds do not overlap and can be identified individually. to obtain the distribution of virtual sound sources.

In the measurement using the four-point method, as shown in FIG. 3, the impulse response from the sound source 26 is measured at four very close sound receiving points 0°x, y, and z within the target acoustic space 24. These sound receiving points O,X.

It is a necessary condition that y and z are not on one plane, but in order to facilitate later processing, one sound receiving point O is used as the reference origin, and the other three are Sound receiving point x, y,
Arrange so that z forms Cartesian coordinates. The distance from the origin 0 is equal to d.

This will be explained using a simple experiment in which a single reflector is installed in an anechoic chamber.

Microphone M r at each sound receiving point o, x, y, z
The outputs of C, M r Cx, M I C, , M I C2° are as shown in FIG. This means that direct sound is transmitted from each microphone MIC, MICx.

Time too at MIC and MIC7, respectively.

tx , ty , tz , and the reflected sound from the reflector is to , tx , ty , tzl
1 1 1.

FIG. 6 schematically shows the path of reflected sound.

Since the sound source 26 is located in the y-axis direction when viewed from the sound receiving points o, x, y, and z, the direct sound first enters the microphone MICy, and then almost simultaneously enters the microphones MIGo, MICx, and MIC2. Therefore, as shown in Figure 5, t y
o < t o ~ t x o field tz .

holds true for direct sound, and holds true for reflected sound.

The distances ro, rxl, ryl, and rzl from each sound receiving point o, x, y, and z to the virtual sound source 26' are expressed by the following equations, where V is the speed of sound.

ro1=v-tol r　X　1””V・tx.

ryl-v@ty.

rz1=v-tzl The coordinates of an arbitrary virtual sound source are (x, Yo, Zo), and the distances from the virtual sound source to each sound receiving point o, x, y, z are ro, rx, ryo, respectively.

n rzo, the equation of a sphere centered at each sound receiving point o, x, y, z and having a virtual sound source on its surface is Xn + Yn
10Zn2=ro 2(Xn-d) +Yn+
Zn2=rx 2Xn 10 (Yn-d) +Zn
2=ryo2Xn +Yn+(Zn-d)2=
It becomes rz 2. Solving this equation (and, d + ro -rxn

In the manner described above, the coordinates of the virtual sound source corresponding to each reflected sound can be determined.

In general, an impulse response is not as simple as shown in Figure 5, but has a complex shape with many reflected sounds gathered together. Short interval cross-correlation is used to select the specific reflected sound produced or peak from the impulse response of each sound receiving point. In other words, the section where the cross-correlation coefficient is the largest with the section where the microphone MICo has the output is set to the section where the microphone MICo has the output.
Select from among the outputs of Cx, MIC, MIC7 and find the arrival time to of the reflected sound.

Determine tx, ty, and tzo.

nn An example of measuring the virtual sound source distribution of a certain hall using the four-point method described above is shown in FIGS. 7 to 9. Figure 7 is X-Y
FIG. 8 is a projection view onto a plane (horizontal plane), FIG. 8 is a projection view onto the Y-Z plane, and FIG. 9 is a projection view onto the X-7 plane. The size of O in the figure represents the level of reflected sound, which is typically measured by the microphone MIC, for example.

■ Method for determining a virtual sound source by calculation A method for determining a virtual sound source by calculation rather than measurement is the mirror image method. As shown in Figure 10, this
4 is compared to a mirror, the sound is emitted from the actual sound source 27, and the sound receiving point 28
When a sound is received in the acoustic space, the reflected sound on the wall surface 24 is regarded as being virtually emitted from the sound FA 30 located at the virtual image position in the mirror, and these virtual sound sources 30 are determined according to the wall shape of the acoustic space. It's something that will happen.

An example of finding the virtual sound source distribution of a certain hall using the mirror image method is shown in FIGS. 11 and 12. FIG. 11 is a projection view onto the X-Y plane (horizontal plane), and FIG. 12 is a projection view onto the Y-Z plane. In the case of the mirror image method, the amplitude level is set depending on the distance from the sound receiving point 28 to the virtual sound source.

When simulating the reflected sound of a performance from each virtual sound source in a room based on the virtual sound source distribution data obtained by measurement or calculation as described above, multiple speakers are placed on all sides of the room, The performance is picked up by a microphone, and the picked-up signal is sent to an appropriate speaker (corresponding to the direction of the virtual sound source) with an appropriate time delay (corresponding to the distance to the virtual sound source) and an appropriate volume (amplitude level of reflected sound). ), it is possible to simulate the reflected sound of a performance.

In this case, the direction, distance, and level of the reflected sound heard at the performance position will vary depending on the performance position and the position of each speaker in the room, so the position (direction and, if necessary, distance) of the speakers relative to the performance position must also be considered. Then, in which direction the reflected sound is emitted from the speaker at what volume and delay time.

Additionally, speakers should ideally be placed in the direction of all virtual sound sources. However, in order to achieve this, speakers would have to be placed throughout at least the upper hemisphere of the room centered around the performance position, which is impossible in reality. Economically, the limit is about 4 to 10 speakers, so arrange that many speakers around the room, define the area each speaker is responsible for, and reflect the virtual sound source included in each area. To simulate each sound by representing it with a corresponding speaker. According to this method, the anti-OA sound from the virtual sound source located between adjacent speakers is representatively emitted by one of the speakers, so strictly speaking, the direction of the virtual sound source is accurately simulated. Although this is not a problem, there is no practical problem as long as the number of speakers is large enough, and considering that there is a limit to the ability of human hearing to determine direction, this is sufficient.

Alternatively, if it is necessary to accurately simulate the direction of a virtual sound source between adjacent speakers, this can be achieved by volume distribution between the speakers.

In this way, when simulating the reflected sound from a virtual sound source located between the speakers by using one speaker as a representative, and when simulating by distributing the volume between the speakers, the volume that should be emitted from each speaker and the Each delay time will be explained.

■ When simulating using one speaker as a representative Figure 13 shows the simulation using eight speakers SP centered on the performance position 1W34.
1 to SP8 are arranged. Here, the acoustic space is divided by a line connecting the central position of the adjacent speakers and the performance position 34, and divided into eight areas d1 to d8 on the horizontal plane. Letting the reflected sound in each area d1 to d8 be PHn,
Reproduction sound PH3 (M-1 to 8) of each speaker SP1 to SP8 necessary to obtain these reflected sounds P)lo at the performance position 34
is expressed by the following equation.

However, NM (M = 1 to 8): Number of virtual sound sources in each region d1 to d8 (-Number of reflected sounds U: Unit function t: Time τ. Delay time of two 04 sounds ■ Volume distribution between adjacent speakers When simulating by

As shown in FIG. 14, four speakers SP1 to SP4 are arranged, for example, in the four corners of the room 36, and lines connecting the performance position 38 and each speaker SP1 to SP4 are used to divide the four quadrants n, m,
1 and k, and each speaker SP1 to SP4 simulates anti-DJ sound from a virtual sound source in the left and right quadrants. That is, speaker SP4. The reflected sound in quadrant n is simulated using the volume ratio of SPI, and speaker SP1
．． The sound m ratio of SP2 simulates the reflected sound in quadrant m, the sound m ratio of speaker SP2° SF3 simulates the reflected sound in quadrant 1, and the sound m ratio of speaker SP2°SF3 simulates the reflected sound in quadrant m. Simulate the reflected sound within the quadrant with the volume ratio of SP4. The reproduced sounds P, .

However, pn, Pm, pl, Pk: Reflected sound τn, τm,
τ1. τ is the delay time θn, θm, τ1. τ = direction angle θ1 of reflected sound on the X-Y plane (horizontal plane). θ, θ, θ: Subika spi ~ X of SP4
- Direction angle τHn' on the Y plane Delay time of each speaker reproduction sound τHm, τH+, τHk'. The operator's correction term is a correction based on the performer's interaural distance, and here we assume a case of 15 CIR.

N I, M I, L l, K I: each quadrant n
, m, I.

The number of virtual sound sources in k is 2 hours U: Unit function Note that in the above equation, in order to simulate the direction of reflected sound from a virtual sound source located between adjacent speakers, the signal distribution between them is calculated as shown in Fig. 15. Although the CO8 function shown in Figure 16 is used, it is recommended to use the linear function shown in Figure 16 or the log function shown in Figure 17, which can best approximate the reflection direction depending on the speaker arrangement, speaker characteristics, etc. .

With the signal distribution described above, reflected sound from all directions can be simulated using the speaker arrangement shown in FIG. 14.

In addition, when performing in an actual hall, the performer will listen to the reflected sound while performing at the performance position, so the sound source and sound receiving point are both set at the performance position, that is, on the stage, and the virtual Once the sound source distribution is determined, if reflected sound is generated using the reflected sound parameters of each speaker based on the virtual sound source distribution, the performer can perform in a normal room with the atmosphere of being on a stage in the hall or other venue. can. Furthermore, the present invention is not limited to this, and by obtaining reflected sound parameters based on virtual sound source distributions obtained by varying the sound source and sound receiving point, it is possible to enjoy a variety of performances.

In addition, in a room, there is a distance between the speakers and the performance position, resulting in a time difference, so to more accurately simulate the reflected sound emitted from the virtual sound source, take this time delay into consideration when setting each speaker. Request playback sound from the speaker.

Figure 18 shows the measurement of reflected sound data (direction, distance, amplitude level) from a virtual sound source in a hall using the four-point method.
Based on this, when simulating reflected sound with the speaker arrangement shown in Figure 4, the signal PH8 (M-1 to 4) to be reproduced from each speaker SP1 to SP4 is calculated from the above equation (2). This shows the reflected sound structure of the signals output from each speaker SP1 to SP4, and can also be considered as an impulse response in each speaker direction. The impulse responses of adjacent speakers are related to each other, that is, the amplitude level delay time of both impulse responses is calculated in advance so that the reflected sound located between these speaker directions is correctly localized in that direction by both speakers. has been done.

When generating reflected sound for a source signal (a continuous signal such as a record playback signal), these impulse responses are calculated using the parameter (
Regarding gain and delay time), the delay time for generating each reflected sound is counted based on the time when the sample value was obtained to generate the reflected sound sequence, and the delay time for each reflected sound is calculated based on the time when the sample value is obtained. Determine the level of the anti-04 sound. ), if these reflected sound sequences obtained for each sample value are added together at each point in time, reflected sounds in each speaker direction are generated, and if these are emitted from the corresponding speakers, the performance position 38 (No. The four performers in Figure 14) can enjoy the atmosphere as if they were playing in a hall with that virtual sound source distribution.

As the reflected sound generation processing based on the reflected sound parameters of the impulse response, a method using convolution calculation, which will be described later, etc. can be used.

In addition, in this invention, the number of microphones is not limited to one, but a plurality of microphones can be used.

Further, it is desirable that sound collection by a microphone is not affected by the acoustic characteristics of the room as much as possible.

To this end, it is desirable to minimize reflections in the room itself, and this can be achieved by applying appropriate sound absorption treatments.

Further, in the present invention, since the sound picked up by the microphone is subjected to signal processing and reproduced again by the speaker, care must be taken not to cause howling. The arrangement of speakers and microphones for this purpose will be described in Examples to be described later.

By the way, the reflected sound from a hall etc. usually lasts about 2 to 3 seconds, so if all of this is to be generated using reflected sound parameters, it is necessary to store performance sound data for that length in memory. In some cases, the memory capacity becomes enormous. However, the initial reflection part (several hundred milliseconds in a general hall)
If only the sound is generated, the sound will suddenly cut off and it will not be possible to sufficiently create the atmosphere of a hall or the like.

Therefore, by using a reverberation adding device consisting of a loop (simple repeated feedback such as a rhombic filter or all-bus filter) to generate the middle to late parts of the reflected sound, the memory capacity is reduced. You can get the sound.Reflections in the actual hall etc. can be clearly recognized as reflected sounds in the early stages, but the later on, the more reflected sounds appear, so it becomes more and more chaotic.
Each line becomes unclear. In other words, the chaotic reflected sound collection that follows is a diffuse sound and is a statistical r, so to reproduce it, it is sufficient to control that O, and use a comb filter or an all-bass filter. A simple repeating type reverberation adding device such as a filter is sufficient.

FIG. 1 is a conceptual diagram showing a sound collection and reproduction control device of the present invention that utilizes the above principle. Here, the direction, delay time (corresponding to distance), and amplitude level of the reflected sound are determined based on the virtual sound source distribution as "reflected sound data" of the hall to be reproduced (for example, Carnegie Hall), etc., and these are From the reflected sound data and the speaker arrangement used for simulation, the delay time and amplitude level of the reflected sound that should be output from each speaker in order to simulate the reflected sound are determined for each speaker as "reflected sound parameters". Control factor 42 for reflected sound parameters
, the processor 46 generates a reflected sound signal to be reproduced by each speaker in order to simulate the reflected sound of the sound (performance sound picked up by a microphone) 44 in each virtual sound source, and the reflected sound signal generated for each of these speakers is Anti-I)
By supplying a single sound signal to speakers 56.58.60.62 via amplifiers 48.50 and 52.54, reflected sound from each virtual sound source is simulated.

[Example 1] FIG. 19 shows an embodiment of the present invention.

The room 80 is subjected to sound absorption treatment to give it dead characteristics. Speakers 56, 58, 60, and 62 are arranged at the four corners of the room 80, facing toward the performer 88. Also, in the room 80, there are five microphones 81, 82, 83 .
84.85 are placed. The microphone 81 is arranged facing the musical instrument 9o, and the microphones 82 to 85 are arranged at intermediate positions between the subtraction forces of 56.58 and 60.62, respectively. Incidentally, the microphone 81 picks up the sound of the instrument itself, and the microphones 82, 83, 84, 85 pick up the sound taking into consideration the directionality of the instrument (which direction the instrument is facing). Note that the practical number of microphones is about 1 to 9.

When the performer 88 plays the musical instrument 90, the sound is collected by the microphones 81 to 85, and these sound signals are mixed by a microphone mixing circuit 92 having a built-in microphone head-up. Then, the processor 46 generates reflected sound based on the reflected sound parameters. These generated reflected sounds are emitted from each speaker 56, 58, 60, and 62 via a 4-channel amplifier 72 (combined amplifiers 48, 50, and 52, 54 in FIG. 1), and are transmitted to the performer. 88 allows you to enjoy the atmosphere as if you were actually in the hall playing.

The reflected sound parameters can be adjusted by the player 88 at his or her performance position by operating the remote controller 76.

A specific example of the processor 46 in FIG. 19 is shown in FIG. 20. In FIG. 20, performance signals picked up by microphones 81 to 85 are mixed by a microphone mixing circuit 100, and the level is adjusted by an input volume 102.

The signal is then A/D converted by an A/D converter 106 via a low pass filter (for preventing aliasing during A/D conversion) and a sample/hold circuit 104. Furthermore, in order to give frequency characteristics to the reflected sound, a digital filter 108°110.112.1 is applied to each channel.
Passed to 14.

The performance signals output from the digital filters 108, 110, 112° 114 are input to reflected sound generation circuits 116, 118, 120° 122 of each channel. The reflected sound generation circuits 116, 118, 120, and 122 generate performance signals for each channel based on the reflected sound parameters (delay time data and gain data) of each channel stored in the memory 126 according to instructions from the microcomputer 124. generate anti-rJia signals, respectively. These generated reflected sound signals are subjected to D/A conversion in a time division multiplexed manner in the D/A converter 124. The output signal of the D/A converter 124 is distributed to each channel in a time-division manner, smoothed through sample-and-hold circuits and low-pass filters 126, 128, 130, and 132, and returned to an analog signal. . The signals are then supplied to each channel speaker 56.58, 60.62 via output volume 134, 136, 138.140 and power amplifier 48, 50, 52.54, respectively. This allows each channel speaker 56.58.60
．． 62, the reflected sounds of the performance from the virtual sound sources in each corresponding direction are generated, and an acoustic space such as a hall specified by the distribution of the virtual sound sources is reproduced.

Note that the memory (ROM) 126 stores reflected sound parameters of various acoustic spaces such as halls and the digital filter 10.
Parameters of frequency characteristics of 8, 110, 112, and 114 are stored for each channel, and based on the operation of the wireless remote controller 76, the microcomputer 124 commands the microcomputer 124 via the remote controller sensor interface 142 to select one of the holes. Parameters are read.

The read frequency characteristic parameters are once transferred to the RAM, and the frequency characteristics of the digital filters 108, 110, 112, and 114 are controlled based on the parameters held in the RAM.

The frequency characteristic parameters held in the RAM can be adjusted according to preference by operating the wireless remote controller 76.

Also, the reflected sound parameters of each channel read out (
delay time data and gain data) are once transferred to the RAM (parameter memory 16o in FIG. 23, which will be described later) provided in the reflected sound generation circuits 116, 118, 120, and 122 of each channel, and are held in this RAM. Based on the reflected sound parameters, the reflected sound generation circuit 116,1
Reflected sound of the performance signal is generated for each channel at 18°120.122. The reflected sound parameters held in the RAM can be finely adjusted by operating the wireless remote controller 76, and thereby the reverberation feeling can be changed according to one's preference.

By the way, the reflected sound generation circuit 116, 118°120.1
22 can generate a reflected sound signal by superimposing (convolution operation F[>) signals obtained by delaying these input signals (performance signals).Reflection sound generation by this convolution operation will be explained below.

Reflected sound generation by convolution calculation involves creating signals with various time delays and amplitude levels from the performance signal based on the reflected sound parameter string of each channel shown in FIG. 18, and superimposing them. That is, to explain one channel, the reflected sound parameter sequence to be used in that channel is determined by the delay time τ, (+-1°■ 2, ..., n) and gain (amplitude level) gi, as shown in FIG.
63, delay signals are taken out from each tap corresponding to the delay time τ·, gain q is applied to them by amplitude adjusters 152-1 to 152-n, and the signals are combined by an adder 153. As a result, from the adder 153, X −Σ
・・gi (3) out 1 i=1 A reflected sound signal is output.

Reflected sound generation circuit 116 (118°120.1
22) is shown in FIG. 23.

Regarding the configuration of the delay memory 163, in the case of analog signals, it is possible to use a charge transfer element such as BBD or COD, and in the case of digital signals, it is possible to use a digital memory that is program-controlled using a shift register or RAM. However, in the following example, there is a large degree of freedom in terms of configuration, and the parameters (delay time and gain)
A case will be explained in which a RAM is used, which is easy to set and change.

In FIG. 23, parameter memory (RAM>160
is stored in the memory ROM by operating the wireless remote controller 76.
126 (Figure 20), the corresponding channel (reflected sound generation circuit 116)
If so, the reflected sound parameters of the front left channel) are stored in each address.

The stored reflected sound parameters are shown in the table below.

In addition, in this table, τ. indicates one sampling period of the input signal. Therefore, the delay time data □, □, ・
...(integer value) is delay τ0 τ0 time τ1. τ2. The position of the sample corresponding to (
(i.e., how many previous samples) it yells.

A data memory (delay memory) 163 is composed of a RAM, and is connected to the digital filters 108, 110, 112.
, 114 (Fig. 20) are sequentially written, and the delay data at positions corresponding to the delay time data rl °1 □, □, . . . stored in the parameter memory are sequentially written. 70′
0 Yu is read out.

The counter 164 indicates the current address to be written in the data memory 163, and is incremented every sample period of the input signal.

The counter 165 specifies the read address of the parameter memory, and counts up from O to n within one sample period of the input signal to read each parameter of delay time data and gain data.

The multiplexer 166 switches the address command applied to the parameter memory 160 to either an input address from the parameter controller 162 or a read address from the counter 165.

The subtracter 167 outputs a value obtained by subtracting the current address from the counter 164 and the delay time data from the parameter memory 160 as an address command for the data memory 163. The data memory 163 is the parameter memory 1
When the read address of 60 is O (that is, when both delay time data and gain data are read as O), the mode is switched to the write mode, so at this time the subtracter 167
The output of (ie, the current address from counter 164) is applied to data memory 163 as a write address command. Further, the data memory 163 includes the parameter memory 16
When the read address of 0 is other than 0, the mode is switched to read mode.
111 address) is added to the data memory 163 as a read address command.

Multiplier 168 provides the delayed signal read from data memory 163 with corresponding gain data currently read from parameter memory 160.

The accumulator 169 performs an elemental ring (convolution operation) on the delayed signal output from the multiplier 168 using a register 175 and an adder 170 to create the reflected sound signal shown in equation (3) above. The reflected sound signal created by the accumulator 169 is then transferred to the D/A converter 124 (FIG. 20).
It is D/A converted and output. In addition, the AND circuit 17
No. 1 is such that every time a reflected sound signal is created, the accumulated data up to that point is interrupted by the signal C3, and the accumulated value is reset to O.

The timing controller 172 is for creating each timing signal 01-C5 for operating each of the above-mentioned circuits.

Next, the operation of the apparatus shown in FIG. 23 will be explained.

(1) Setting reflected sound parameters First, the hole to be reproduced is selected by operating the wireless remote controller 76. As a result, memory 1
26 (FIG. 20), the reflected sound parameters τ to τ, 01 to ngo of the corresponding hall are read out and written into the parameter memory 160. Written reflected sound parameters τ1~τ, Q1
~C+o can be adjusted by operating the wireless remote controller 76.

When performing this writing and adjustment, the parameter memory 160 is switched to the write mode and the multiplexer 166 is switched to the parameter address 166 side.

(2) After setting the reverberation signal creation parameters, set the parameters in the parameter memory 16.
0 is switched to the read mode, the multiplexer 166 is switched to the counter 65 side, and an input signal (a microphone pickup signal) is supplied to the data memory 163 to create a reflected sound signal.

To create a reflected sound signal, one sampling period of the input signal is
As a unit, among them: (1) writing an input signal to the data memory 163; and (2) each delay time τ1 to τ set from the data memory 163. 2. Reading out the delayed signal corresponding to 2) Weighting (g1 to qn) for each of the read delayed signals 2. Performing accumulation to create a reflected sound signal. Each step will be explained with reference to timer 1 in FIG. 24.

(2) Writing of input signal to data memory 163 At the rising edge of clock C1, clock C4 becomes low level and counter 165 is cleared. Therefore, the parameter memory 160 is designated with address O, and the delay time data,
"0" is read out for both the gain data. Then, at the next rising edge of the clock C5, the clock C2 also rises, and the data memory 163 enters the write state. At this time, the delay time data from the parameter memory 160 is read out as rO as described above.
J, the output of the subtracter 167 is the output of the counter 164 itself, and the input signal is written to the address indicated by the output of the counter 164 in the data memory 163.

■ Reading the delayed signal from the data memory 163 When the data memory 163 has been occupied, the data memory 163
63 is in read mode. Clock C5 performs the above writing once and reading n times within 11 numbering periods.
It rises a total of n+i times. The counter 165 counts this clock C5, adds the count value to the parameter memory 160, and reads each parameter τ to τ and g1 to go of delay time data and gain data.

For example, when the counter value of the counter 165 is "1", the delay time data □ and the gain data OQl are read from address 1 of the parameter memory 160. Furthermore, sequentially from address 2 to τ2
τn and g2. ...from address n to τ0 τ0 and g. are read out respectively.

The delay time data read from the parameter memory 160 is subtracted from the count value of the counter 164 by a subtracter 167, and the subtracter 167 outputs the count value of the counter 164, that is, the delay time data based on the current address. The address before the indicated distance is output, and the address is stored in the data memory 16.
Delay signal stored in the corresponding address from 3 x 1
~x.

is read out.

■ Weighting is such that the delayed signal read from the data memory 163 is multiplied by 1
In the device 168, each corresponding gain data g1 to go read out from the parameter memory 160 is provided.

(2) The accumulated reflected sound signal is obtained by accumulating the data G''X1 to 'nnl.X output from the multiplier 168 while the count value of the counter 165 changes from "1" to rnJ. To perform this accumulation, when the count value of the counter 165 is "1", the clock C3 falls and the previous accumulated value of the accumulator 169 is reset. That is, when the counter 165 is "1", the AND circuit 171 is turned off, and the output of the adder 170 becomes the value of only the output Q''X1 of the multiplier 168 and is held in the register 75. At the timing, the register 175 outputs Q −X, and the adder 1
At 70, it is added to the next data g -x, and the value of register 175 is rewritten.Thus, the addition (accumulation) is repeated in sequence, and 0 terms are added to obtain ΣQ, ·X. By the way, this value is output as a reflection i=1 sound signal.

Through the above operations, a reflected sound signal is generated for each sampling period of the input signal (performance signal). In addition,
In the above explanation, one of the multiple channels of reflected sound signals
Although only one channel is shown, reflected sound signals of other channels can be generated with a similar configuration.

After reading out the reflected sound parameters of the ball from the parameter memory (RAM) 160 (Fig. 23) from the EIJ (ROM) 126 (Fig. 20), the parameters can be adjusted using the wireless remote control 76.
Reflection properties can be changed.

Examples of adjustments include the following:

Figure 25 shows the delay time for speaker 1.2 among the reflected sound parameters of each channel shown in Figure 18, multiplied by a coefficient to relatively expand or reduce the delay time. It is.

This corresponds to varying the width of the hole to be reproduced (5IZE), and if you multiply the delay time by a large ≧ coefficient (>1), the hole will feel wider, and a small coefficient ( If you multiply 1) and shorten the delay time, the hole will feel narrower.

In this way, the feeling of expansion of the hole can be adjusted to approximately 0.0 to 3.0 times (by increasing the number of memory 8s, it is possible to increase it to any multiple).

In FIG. 26, the slope of the gain (corresponding to the amplitude level of the reflected sound) of the reflected sound parameter sequence is varied, and the live feeling (LIVENESS) is thereby varied. That is, if the gain slope is made steep, it will give a dead characteristic, and if it is made gentle, it will give a live characteristic. This is achieved by increasing or decreasing the level of the nearby reflected sound. It is also possible to periodically change the delay time gain of the reflected sound parameter sequence or both of them. For example, when each parameter value is varied using a sinusoidal low-frequency signal, the spatial clarity of the reproduced acoustic space becomes blurred (DIF "tlS").
ION), and it is also possible to obtain special sound effects.

Here, another example of arrangement of speakers and microphones will be explained.

■ Arrangement in FIG. 27 FIG. 27 shows an arrangement row of four speakers and four microphones. That is, the speakers 56.58.60.62 are placed at the four corners of the ceiling of the room 80, and the microphones 82.62.
83, 84.85 are each ceiling speaker 56.58.60
．． 62, respectively. Each microphone 82, 83, 84, 85 has an adjacent speaker 56
．． Since the microphones 82, 83 and 84.85 are located at the same distance from 5B and 60.62, howling is not likely to occur.In addition, the microphones 82, 83 and 84.85 are mounted on the ceiling wall, and the frequency characteristics of the microphone input are as follows. is arranged so that it is equal to 1■.

■ Arrangement in FIG. 28 In FIG. 28, microphones 82, 83, 84° 85 are arranged at the four corners of the floor of a room 80. In this arrangement, the sound pressure at the microphone pickup point is the highest over the entire frequency band, and the frequency characteristics are flat. In addition, the speaker 56,
The influence of direct sound radiation from 58.60°62 is small and the howling margin is large.

■ Arrangement in Figure 29 Figure 29 shows four directional microphones 82°83.8
4.85 near the four corners of the ceiling in the direction of the instrument sound source (room 8
0 center direction). Microphones 82, 83° 84.85 have a loudness of 56, 58
．． Although it is located close to the radiation axis of 60°62, it has strong directivity, and the speakers 56, 58, 60°62
Since I have my back to the speaker, the speaker is 56.58°60.62
No sound is picked up, and the howling margin increases significantly.

[Example 2] In the example shown in FIG. 30, microbons 82.83°84.
85 collected sound signals are individually processed. That is, the sound signals from each of the microphones 82, 83, 84.85 are input to A via input volumes 140 to 143, low pass filters and sample and hold circuits 144 to 147.
A/D conversion is performed by /D converters 148 to 151. A/D
The converted signals of each channel are input to digital filters 108, 110, 112, and 114, respectively.
Thereafter, reflected sound is generated in the same manner as in the embodiment shown in FIG. 20 above. In this way, since each microphone pickup signal is independent, it is also possible to control the sense of direction of the musical instrument indoors. Particularly for directional sound sources, a more natural hall feeling can be reproduced at the playing position.

In this case, if the arrangement of the speakers and microphones shown in FIG. 29 is used, the sound is collected in each direction of the speakers and reproduced from the speakers, which facilitates signal processing of the directional characteristics of the sound.

[Embodiment 3] By the way, in the reflected sound generation using the reflected sound parameters of the present invention, in order to faithfully reproduce the reflected sound from a hall, etc., a memory capacity R corresponding to the duration of the reflected sound is required, so it takes a long time. If you try to measure the reverberation time, the memory will become enormous.

The embodiment shown in FIG. 31 solves this problem and allows a long reverberation time to be obtained with a small memory of 8 memories.

In other words, the early reflected sound is generated individually one by one in the reflected sound generation process using the reflected sound parameters mentioned above, and for the middle to late reflected sound, which is mostly diffused, simple filters such as comb filters and all-pass filters are used. The reverberation sound is generated in the process of generating reverberant sound by using a reverberation adding device that can be configured with a repeating loop.

In FIG. 31, the reflected sound generation circuit 116°118.1
20 and 122 are constructed similarly to those used in the embodiments shown in FIGS. 20 and 30, and generate early reflected sounds from each channel input signal. In addition, the reverberation sound generation circuits 186, 188°190.192 are composed of comb filters, all-bus filters, etc., and each channel input signal or reflected sound generation circuit 116, 118, 120°12
The middle to late reflection sound is generated from the output signal of No. 2. The output signals of the reflected sound generation circuits 116, 118゜120.122 and the reverberant sound generation circuit 186゜188.190.192 are synthesized for each channel, and through the same signal processing as in the previous embodiment, are output to each speaker 56. ,58,60
．． 62.

The reflected sound characteristics (output signal of each channel when an impulse is input) obtained in the example of FIG.
An example is shown in FIG. The early reflected sound part is obtained by the reflected sound generation circuit 116 (118, 120° 122), and the middle to late reflected sound part is obtained by the reverberant sound generation circuit 186 (188, 190,
192).

Note that the reflected sound generation circuits 116, 118, 120° 122
If the memory capacity ω allows, the reflected sound generation circuit 116.118
, 120, 122 may have early to middle reflection portions, and the residual W sound generation circuit 186, 188, 190, and 192 may have late reflection portions.

In addition, reflected sound generation circuits 116, 118, 120° 122
The supporting area and the residual W sound generation circuit 186°188.190,
It is not necessary to completely divide the 192 receiving areas, and they may be overlapped at a joint.

Specific examples of the reflected sound generation circuit 116 (118, 120° 122) and the reverberant sound generation circuit 186 (188, 190° 192) are shown in FIG.

The reflected sound generation circuit 116 has the same configuration as that shown in FIG. 22, and uses a multi-type delay memory 163 to take out delayed signals from each tap corresponding to the delay time τi. , amplitude adjuster 1
A gain g1 is applied to each of the signals 52-1 to 152-n, and the adder 153 synthesizes them. As a result, the adder 15
3 outputs a reflected sound signal as follows.

The delay memory 63 is provided with a tap corresponding to a delay time τX longer than the delay time τn, and the delayed signal from this tap is given a gain gx by an amplitude adjuster 152-X and input to the reverberation sound generation circuit 186. has been done.

The reverberation sound generation circuit 186 is composed of an all-pass filter. That is, the reverberation sound generation circuit 186 inputs the input signal to the delay circuit 196 via the adder 194, and feeds back the output of the delay circuit 196 to the adder 194 via the amplifier 198 (gain 1-02). . The output of the adder 194 is sent to the adder 202 via the amplifier 200 (gain -〇).
is added to the output of the delay circuit 196. In this way, the adder 194 outputs a reverberation signal.

The reflected sound signal output from the reflected sound generation circuit 116 and the reverberation sound signal output from the reverberation sound generation circuit 186 are sent to the adder 2.
04 and output. The reflected sound generation circuit 116 outputs reflected sounds with delay times τ1 to τn, and the reverberant sound generation circuit 186 outputs reflected sounds with delay times τ slower than τn.
Since the reverberant sound after X is output, the adder 204 outputs a series of long reflection signals that connect the two.

Figure 34 gives the impression that the duration of the reflection has been expanded in the example of Figure 31 (approximately twice as long as in the case of Figure 32), and the width of the hole (5IZE) has been expanded. It was designed so that

This is reflected sound generation circuit 116, 118, 120°12
2 and the delay time parameters in the reverberation sound generation circuits 186, 188, 190, and 192 are multiplied by a coefficient (~2).

[Embodiment 4] In the embodiment shown in FIG. 35, the remaining sound 51 is applied all at once before the reflected sound is generated for each channel 17.

In other words, the mixed microphone sound signal is A/D
After Δ/D conversion by the converter 106, the reverberant sound generation circuit 21
Reverberation sound is added to all the signals at 0, and then input to reflected sound generation circuits 116, 118° 120, 122 via digital filters 108, 110, 112° 114, and reflected sounds are generated for each channel. . With such a configuration, it is not necessary to prepare a reverberation sound generation circuit for each channel, so the configuration is simplified.

〔Effect of the invention〕

As explained above, according to the present invention, the anti-rJ4 sound data of a hall, etc. is obtained from the virtual sound source distribution of the hall, etc., and from this reflected sound data and the speaker arrangement, each The reflected sound parameters for generating the reflected sound to be emitted from the speaker were determined, and the reflected sound of the microphone sound signal was generated based on the reflected sound parameter and emitted from the speaker. In the rooms, test chambers, etc., you can enjoy or practice playing musical instruments or singing in an atmosphere as if you were in a large hall.

[Brief explanation of drawings]

FIG. 1 is a conceptual diagram of a sound collection and reproduction control device according to the present invention. FIG. 2 is a conceptual diagram of a conventional sound control device. FIG. 3 is a perspective view showing a method for measuring a virtual sound source using the four-point method. FIG. 4 is a perspective view showing the arrangement of the microphones in FIG. 3. FIG. 5 is a waveform diagram showing the measurement results of the impulse response by the microphone of FIG. 4. FIG. 6 is a diagram showing a method of calculating a virtual sound source position based on the measurement results of FIG. 5. Figures 7, 8, and 9 are diagrams showing the virtual sound source distribution obtained by the four-point method, and Figure 7 is an X-Y plane projection diagram, and Figure 8 is a
The figure is a YZ plane projection view, and FIG. 9 is an xZ plane projection view. FIG. 10 is a diagram showing the principle of the virtual sound source measurement method using the mirror image method. Figures 11 and 12 are diagrams showing the virtual sound source distribution obtained by the mirror image method.
The figure is a YZ plane projection view. FIG. 13 is a plan view showing how reflected sound is reproduced by eight surrounding speakers. FIG. 14 is a plan view showing a reflected sound reproduction state by four surrounding speakers. Figures 15, 16, and 17 show the volume distribution between each speaker in order to simulate reflected sound between adjacent speakers, and Figure 1.5 shows C08l.
Figure 16 is based on a linear function, and Figure 17 is based on a log function. Figure 18 shows a reflected sound parameter string used to create the reflected sound to be supplied to each speaker when simulating the reflected sound with the speaker arrangement shown in Figure 14 based on reflected sound measurement data using the 4-point method. FIG. FIG. 19 is a block diagram showing one embodiment of the present invention. FIG. 20 is a block diagram showing an example of the configuration of the processor 46 in FIG. 19. Figure 21 shows the reflected sound generation circuit 116°118 of Figure 20.
．． 120.122 is a diagram that does not show a reflected sound parameter sequence used for reflected sound generation. FIG. 22 shows the reflected sound generation circuit 116 (118) of FIG. 20 configured to generate a reflected sound signal of the input signal by convolution using the reflected sound parameters of FIG.
, 120, 122). FIG. 23 shows the reflected sound generation circuit 116 (118) of FIG.
, 120, 122). FIG. 24 shows the operation of the circuit of FIG. 23 at time 1?
- It is. FIG. 25 is a reflected sound parameter sequence showing an example in which the delay time of the reflected sound parameter is multiplied by a coefficient to adjust the perception of the size of the hall. FIG. 26 is a reflected sound parameter sequence showing an example in which the live feeling is adjusted by changing the slope of the reflected sound parameter sequence. FIG. 27 to FIG. 29 are perspective views showing other examples of arrangement of speaker microphones. FIG. 30 is a block diagram showing a second embodiment of the invention. FIG. 31 is a block diagram showing a third embodiment of the invention. FIG. 32 is a diagram showing a reflected sound signal pattern obtained in the embodiment of FIG. 31. FIG. 33 is a block diagram showing a configuration example of the reflected sound generation circuit and the reverberation sound generation circuit in FIG. 31. FIG. 34 is a diagram showing a reflected sound signal pattern in which the perception of the size of the hall is finely adjusted in the embodiment of FIG. 31. FIG. 35 is a block diagram showing a fourth embodiment of the invention. 46... Processor, 56.58.60.62...
Speaker, 80... Room, 81, 82, 83, 84° 85... Microphone, 116, 118, 120° 1
22...Reflected sound generation circuit, 186, 188°190.
192,210... Reverberation sound generation circuit. Fig. 17 7'-ゝ＼ 町吋(■ ＼N--)/ Fig. 10~ 0 Fig. 12 1. -Tzu1-Experiment Be-Be-Be-B-
Interpretation N Ward - Daiichi Beb - Bone N ni - Negative. Written amendment of procedure (release) 1. Indication of the case 1985 Patent Application No. 1016092, Name of the invention Sound collection and playback device 1111j3, Person making the amendment Relationship to the case Patent applicant (407) Nippon Gakki Manufacturing Co., Ltd.4, Agent (postal code) 1o5) 6. Subject of correction

Claims (1)

[Scope of Claims] A plurality of speaker reproduction means disposed around a performance position; a microphone sound collection means for collecting performance sounds; Parameter storage means for storing parameters of reflected sounds to be emitted by each of the speaker reproduction means (reflected sound parameters) in order to reproduce reflected sounds in an acoustic space or a model space similar thereto; and a parameter storage means for storing reflected sound parameters in the parameter storage means. and a reflected sound generating means that generates reflected sounds of the sound signals picked up by the microphone sound collecting means based on each of the reflected sound parameters and supplies them to corresponding ones of the plurality of speaker reproduction means. A sound collection and playback control device featuring: