JP2003263178A - Reverberator, method of reverberation, program, and recording medium - Google Patents

Abstract

<P>PROBLEM TO BE SOLVED: To provide a reverberator capable of expressing a sufficient sound field effect by using a simple configuration. <P>SOLUTION: Data included in a prescribed period from the generation of an impulse sound out of the sampling data of an impulse response waveform for adding a reverberation effect are stored as initial reflected sound data and data included in a succeeding prescribed period are stored as rear reverberation data. An initial acoustic signal is generated by convoluting the initial reflected sound data into the sampling data of an acoustic signal to be processed. A rear acoustic signal is generated by convoluting the rear reverberation dada into the sampling data. A reverberation signal is generated by repeatedly outputting the rear acoustic signal while attenuating the rear acoustic signal. Data obtained by combining the initial acoustic signal, the rear acoustic signal and the reverberation signal are outputted as data to which a sound field effect is imparted. <P>COPYRIGHT: (C)2003,JPO

Description

Detailed Description of the Invention

[0001]

BACKGROUND OF THE INVENTION 1. Field of the Invention The present invention relates to a technique for adding an acoustic space reverberation effect to an acoustic signal.

[0002]

2. Description of the Related Art An impulse response waveform in an acoustic space such as a hall or a church is recorded, and sampling data of this impulse response waveform is convoluted with an acoustic signal so that an initial reflected sound or a reverberant sound in the acoustic space is recorded. Device that gives effect (hereinafter referred to as reverberation device)
There is. The sampling data of the impulse response waveform in the acoustic space includes impulse sound emitted from a sound source installed in the acoustic space and TSP (Time Stretched Pul).
se) and other acoustic measurement signals are collected by a microphone,
After that, it can be obtained by sampling the analog signal waveform of the sound converted into an electric signal and performing processing as necessary.

[0003]

By the way, in an acoustic space having a long reverberation time, the reflected sound and the reverberation sound of the impulse sound are collected even after a while after the generation of the impulse sound. To reproduce the space, the impulse response waveform takes a long time. Then, in order to reproduce such a reverberant space, it is necessary to convolve a large amount of sampling data, which requires a huge amount of hardware resources. Therefore, conventionally,
As shown below, a method with a relatively simple hardware configuration was adopted.

(Method 1) A method of using only the initial part of the sampling data of the impulse response waveform. According to this method, the convolution operation is performed using the data included within the predetermined period from the generation of the impulse sound to generate the signal of the initial reflected sound. The following reverberant sound signal is artificially generated using a recursive filter regardless of the impulse response. Then, a method of synthesizing a signal of a reverberant sound that follows the signal of an initial reflected sound. However, in this method, since the signal of the subsequent reverberation sound is generated independently of the impulse response, there is a problem that when the signal of the subsequent reverberation sound is connected to the early reflection sound, the connection becomes unnatural in terms of hearing. there were. In order to make the signal connection natural, complicated work such as fine adjustment of the filter coefficient is required.

(Method 2) A method of using only main data of the sampling data of the impulse response waveform. For example, of values of data obtained by sampling an impulse response waveform at a predetermined sampling frequency, data having a certain level or higher is extracted and treated as main data, and convolution calculation processing is performed using only this main data. Is. However, this method uses only the main data of the sampling data of the impulse response waveform. That is, since data other than the main data is thinned out, the acoustic characteristics of the space are considerably lost, and there is a problem that a sufficient acoustic space cannot be reproduced.

The present invention has been made in consideration of the above points, and provides a reverberation applying device, a reverberation applying method, a program, and a recording medium capable of expressing a sufficient sound field effect with a simple configuration. The purpose is to do.

[0007]

In order to solve the above-mentioned problems, the reverberation imparting apparatus according to the present invention has a first predetermined time from the time when an impulse sound is generated in an impulse response waveform to an acoustic signal to be processed. A first convolution operation unit for generating an initial acoustic signal by convolving sampling data corresponding to an initial period until a lapse of time, and a second predetermined period from the lapse of the initial period of the impulse response waveform with respect to the acoustic signal. A second convolution calculation unit that generates a rear acoustic signal by convolving the sampling data corresponding to the period after the passage of time, and an attenuation calculation unit that repeatedly outputs the rear acoustic signal while attenuating the rear acoustic signal. And an output unit for combining and outputting the initial acoustic signal, the rear acoustic signal, and the reverberation signal. And it features.

According to the reverberation imparting apparatus having such a configuration, among the sampling data of the impulse response waveform,
An initial acoustic signal (a signal mainly related to an initial reflected sound) can be generated by convolving the sampling data corresponding to the initial period from the time when the impulse sound is generated to the time when the first predetermined time has elapsed (first convolution operation). Part). Further, among the sampling data of the impulse response waveform, it is possible to generate a rear acoustic signal (a signal subsequent to the initial reflected sound) by convolving the sampling data corresponding to the second predetermined period after the initial period has elapsed. Yes (second convolution operation unit). Therefore, the impulse response can be faithfully reproduced for the data relating to the relatively early stage that characterizes the acoustic space. Further, a reverberation signal (a signal related to reverberation sound) can be generated by attenuating a signal obtained by convolution by the second convolution operation unit and repeating it (attenuation operation unit). Therefore, since it is not necessary to convolve all the sampling data of the impulse response waveform, it is possible to take a simple configuration without requiring huge hardware resources. Since the initial acoustic signal, the rear acoustic signal, and the reverberation signal are all generated based on the same impulse response waveform, even when the initial acoustic signal, the rear acoustic signal, and the reverberation signal are combined (output unit), The signal connection will not be unnatural. That is, it is possible to express a sufficient sound field effect with a simple configuration.

Further, the reverberation applying apparatus according to the present invention convolves the acoustic signal to be processed with the sampling data corresponding to the initial period from the time when the impulse sound is generated in the impulse response waveform to the time when the first predetermined time has elapsed. A first convolution operation unit for generating an initial acoustic signal, and convolution of the acoustic signal with sampling data corresponding to the period from the passage of the initial period of the impulse response waveform to the passage of a second predetermined time. A second convolution operation unit for generating a rear acoustic signal, an attenuation operation unit for repeatedly generating an attenuation signal by attenuating the rear acoustic signal, and at least one of the density and the phase of the attenuation signal. And a diffusion unit that generates a reverberation signal by adjusting the initial acoustic signal, the rear acoustic signal, and the reverberation signal. Or it may be characterized by comprising an output unit configured to and outputs, a.

Further, each of the reverberation applying devices described above may have a storage unit for storing sampling data to be convolved in the first convolution operation unit and the second convolution operation unit. Further, it is preferable that the output unit outputs the rear acoustic signal with a delay corresponding to the time length of the sampling data corresponding to the initial period.

In the reverberation applying method according to the present invention, the sampling data corresponding to the initial period from the time when the impulse sound is generated in the impulse response waveform to the time when the first predetermined time elapses is convoluted with the acoustic signal to be processed. A first convolution operation step for generating an initial acoustic signal,
A second convolution calculation step of generating a rear acoustic signal by convolving the acoustic signal with sampling data corresponding to a period from the initial period of the impulse response waveform to the second predetermined period of time; An attenuation calculation step of generating a reverberation signal by repeatedly outputting the rear acoustic signal while attenuating, and an output step of combining and outputting the initial acoustic signal, the rear acoustic signal and the reverberation signal. Is characterized by.

In the reverberation applying method according to the present invention, the sampling data corresponding to the initial period from the generation of the impulse sound in the impulse response waveform to the elapse of the first predetermined time is convoluted with the acoustic signal to be processed. A first convolution operation step for generating an initial acoustic signal,
A second convolution calculation step of generating a rear acoustic signal by convolving the acoustic signal with sampling data corresponding to a period from the initial period of the impulse response waveform to the second predetermined period of time; An attenuation calculation process for generating an attenuation signal by repeatedly outputting the rear acoustic signal while attenuating, a diffusion process for generating a reverberation signal by adjusting at least one of the density and the phase of the attenuation signal, and the initial acoustic A signal, the rear acoustic signal and the reverberation signal are combined and output.

A program according to the present invention causes a computer to convolve sampling data corresponding to an initial period from the time when an impulse sound is generated in an impulse response waveform to the time when a first predetermined time has elapsed with an acoustic signal to be processed. First convolution calculation means for generating an initial acoustic signal by: and convolving the acoustic signal with sampling data corresponding to a period from the time when the initial period of the impulse response waveform elapses to the time when a second predetermined time passes. Second convolution calculation means for generating a rear acoustic signal by the above, attenuation calculation means for generating a reverberation signal by repeatedly outputting while attenuating the rear acoustic signal, the initial acoustic signal, the rear acoustic signal and the reverberation. A program that functions as an output unit that synthesizes and outputs signals And it features.

Further, the program according to the present invention causes a computer to store sampling data corresponding to an initial period from the time when an impulse sound is generated in an impulse response waveform to the time when a first predetermined time elapses with respect to an acoustic signal to be processed. First convolution calculation means for generating an initial acoustic signal by convolving, and sampling data corresponding to the acoustic signal during a period from the passage of the initial period of the impulse response waveform to the passage of a second predetermined time. Second convolution calculation means for generating a rear acoustic signal by convolution, attenuation calculation means for repeatedly generating the rear acoustic signal while attenuating the rear acoustic signal, and at least one of the density and phase of the attenuated signal. Means for adjusting a reverberation signal by adjusting one of the two, and the initial acoustic signal and the rear acoustic signal. Preliminary may be characterized in that the a program for causing a reverberation signal as an output means for combining and outputting.

If the above-mentioned program is recorded in a computer-readable recording medium, it is possible to facilitate the transaction and the like.

[0016]

BEST MODE FOR CARRYING OUT THE INVENTION Next, preferred embodiments of the present invention will be described with reference to the drawings. A: Configuration / Operation of Embodiment FIG. 1 shows an effector 100 according to an embodiment of the present invention.
It is a block diagram which illustrates the structure of. Effector 100
Stores sampling data of impulse response waveforms measured in acoustic spaces such as halls and churches and impulse response waveforms obtained by simulation, and the acoustic data is processed by convoluting the sampling data with acoustic signals. It has a function as a reverberation applying device that generates a signal to which a reverberation effect such as an early reflection sound of space or a rear reverberation sound is added.

The effector 100, as shown in FIG.
Operation unit 101, ROM (Read Only Memory) 102, R
AM (Random Access Memory) 103, A / D (Analog)
/ Digital) conversion circuit 104, CPU (Central Process)
sing unit) 105, display unit 106, reverberation data memory 107, and D / A (Digital / Analog) conversion circuit 10
8. The reverberation adding unit 120 is provided, and the units are connected to each other via the bus 109. The microphone 10 is connected to the A / D conversion circuit 104, and the speaker 40 is connected to the D / A conversion circuit 108 via the amplifier 30.

When the user operates a key on the operation panel, the operation section 101 outputs an operation signal corresponding to the operation to the CPU 105. Various programs for controlling each part of the effector 100 are stored in the ROM 102 in advance. The RAM 103 is used as a working area, and temporarily stores data required for processing such as reverberation addition.

The A / D conversion circuit 104 samples and outputs an input signal every time a sampling clock having a predetermined frequency is applied. The CPU 105 is the ROM 102
By executing the program stored in, each unit of the device connected via the bus 109 is controlled. The display unit 106 includes a liquid crystal display panel and a drive circuit that controls the display of the liquid crystal display panel.

The reverberation data memory 107 stores sampling data of impulse response waveforms. Reverberation data memory 107 of the effector 100 according to the present invention
Stores only a part of all sampling data of the impulse response waveform. FIG. 2 schematically shows sampling data of the impulse response waveform. In FIG. 2, the horizontal axis represents time and the vertical axis represents signal level. The sampling time is Ts. Among the sampling data of the impulse response waveform, the data D1 included in the predetermined period T1 from the time when the impulse sound is generated (for example, the period of 0.3 seconds from the time when the impulse sound is generated) is reverberation data as “data for initial reflected sound”. Memory 107
Stored in. Further, a predetermined period T2 subsequent to the predetermined period T1 (for example, 0.3 to
The data D2 included in the 0.5 second period) is stored in the reverberation data memory 107 as "data for rear reverberation sound". Here, the period T1 and the period T2 are temporally continuous. If there is a section whose data value is almost 0 in the front part of the “data for initial reflected sound”, the data of such section may not be stored in the reverberation data memory 107. As a result, the amount of memory used in the reverberation data memory 107 can be reduced.

As shown in FIG. 3, the reverberation data memory 107 stores the data value of the sampling data in the corresponding period as time series data for each sampling time Ts. As the data value information, the sampling data value itself of the impulse response waveform may be stored, or a value standardized at a certain level of the sampling data of the impulse response waveform may be stored. Further, the time information of each sample data may be stored in association with the data value.

The reverberation adding section 120 has a function of generating data in which a reverberation effect is added to sampling data such as an acoustic signal which is an input signal. FIG. 4 is an internal block diagram of the reverberation imparting unit 120.
Convolution operation unit 121, convolution operation unit 122, adder 123, delay device 124, filter 125, delay device 12
7, an adder 128 is provided.

The convolution operation unit 121 performs an operation process of convolving the sampling data of the acoustic signal with the initial reflected sound data D1 stored in the reverberation data memory 107, and outputs the operation result to the adder 128. As shown in FIG. 5, the convolution operation unit 121 includes a delay unit 121D-1,
121D-2, ..., 121D- (m-1), the multiplier 12
1A-0, 121A-1, 121A-2, ..., 121A
-(M-1), adders 121K-1, 121K-2,
, 121K- (m-1), and performs convolution calculation processing of m stages. Delay devices 121D-1, 121
D−2, ..., 121D− (m−1) delay time T121
Corresponds to the sampling time Ts of the impulse response waveform. In addition, the multipliers 121A-0, 121A-1,
, 121A- (m-1), the initial reflection sound data D1 (La1, La2, ..., Lam) in the reverberation data memory 107 are set. Specifically, the initial reflection sound data La1 is used as the multiplication coefficient of the multiplier 121A-0, the initial reflection sound data La2 is used as the multiplication coefficient of the multiplier 121A-1, ..., The multiplier 121A- (m
The initial reflected sound data Lam is set as the multiplication coefficient of (-1).

The convolution operation unit 122 performs an operation process of convoluting the rear reverberation sound data D2 stored in the reverberation data memory 107 with respect to the sampling data of the acoustic signal, and outputs the operation result to the delay unit 124 and the delay unit 127. To do. As shown in FIG. 6, the convolution operation unit 122 includes delay units 122D-1, 122D-2, ..., 122D- (n.
-1), multipliers 122A-0, 122A-1, 122A
-2, ..., 122A- (n-1), adder 122K-
, 122K- (n-1), and performs n-stage convolution operation processing. Delay device 1
22D-1, 122D-2, ..., 122D- (n-1)
The delay time T122 corresponds to the sampling time Ts of the impulse response waveform. Also, the multiplier 122A-
0, 122A-1, 122A-2, ..., 122A- (n
−1) as the respective multiplication coefficient of the rear reverberation sound data D2 (Lb1, Lb2, ..., Lb) in the reverberation data memory 107.
n) is set. Specifically, the rear reverberation sound data Lb1 and the multiplier 122A are used as the multiplication coefficient of the multiplier 122A-0.
The rear reverberation sound data Lb2, ... Is set as the multiplication coefficient of −1, and the rear reverberation sound data Lan is set as the multiplication coefficient of the multiplier 122A- (n−1).

The delay device 127 delays the data for a predetermined period T127. The value of the predetermined period T127 is adjusted to a value corresponding to the time length of the initial reflected sound data to be convolved in the convolution operation unit 121 (the period T1 is assumed in this embodiment). The delay device 124 has a predetermined period T1.
Delay the data by 24. The value of the predetermined period T124 is adjusted by the convolution operation unit 121 to be larger than the time length of the initial reflected sound data to be convolved, that is, the value of T1 (value of period T127 <value of period T124).

The filter 125 is a filter having a feedback loop, and in the present embodiment, a comb filter (Comb filter) as shown in FIG. 4 is adopted. More specifically, as shown in FIG. 4, the filter 125 includes delay devices 125D and 125ID and a low-pass filter 1.
25L, amplifier 125A, 125GA, adder 125K
125F-1, 125F-
2, ..., 125F-p and P pieces are connected in parallel. Here, the delay device 125ID is a filter 125F-
It plays a role as an initial delay that gives a predetermined delay to one input signal. Also, the amplifier 125G
A adjusts the level of the entire output signal of the filter 125F-1. The low-pass filter may be any filter that attenuates high frequencies, and a shelving filter may be used.

FIG. 7 is a diagram showing an output signal when a pulse signal is supplied to the filter 125F-1. As shown in FIG. 7, the output signal of the filter 125F-1 is a signal in which data is followed at each delay time T125 by the delay device 125D. By adjusting the amplification coefficient of the amplifier 125A to a number less than 1, a signal having a natural attenuation characteristic as shown in FIG.
It is possible to output from 1.

Here, the low-pass filter 125L forming the filter 125F-1 can remove high-frequency signal components of a predetermined frequency or higher. Therefore, the filters 125F-1, 125F-2, ..., 125F-p of each stage
By adjusting the filter coefficient of the low-pass filter that configures the filter, and by adjusting the amplification coefficient of the amplifier 125GA for amplifying the output signal of the filter at each stage, for example, the actual sound that the reverberation time becomes shorter as the signal becomes higher in frequency. It is possible to reproduce the acoustic characteristics in space.

FIG. 8 shows the output signal S1 of the convolution operation unit 122 and the filters 125F-1, 125F-2, ..., 1
25F-p output signals S1-1, S1-2,
, S1-p and the output signal S2 of the filter 125 are schematically shown. Convolution operation unit 122
The signal S1 obtained from the calculation result of
, F, 125F-2, ..., 125F-p. Then, from each filter, a signal component corresponding to the filter coefficient of the filter is repeatedly attenuated and output. Then, the output signals S1-1, S1- of each filter
The combined signal S2 of 2, ..., S1-p is output from the filter 125. That is, as shown in FIG.
Signal S1 obtained from the calculation result of convolution calculation section 122
The attenuation signal S2 having a longer period Tf (Tf> Tc) than that of the period Tc can be generated and output by the filter 125.

For the sake of convenience, each filter 125 is shown in FIG.
Output signal S of F-1, 125F-2, ..., 125F-p
Similar signal types are used for 1-1, S1-2, ..., S1-p. However, in reality, each filter 125F-
The output signals S1-1, S1-2, ..., S1-p depend on the filter coefficient values of 1, 125F-2 ,.
The signal shapes (frequency characteristics and attenuation characteristics) of are different.

The adder 128 and the adder 123 output a combination (addition) of the two supplied signals. The adder 128 outputs a signal obtained by combining the signal obtained from the convolution operation unit 121 and the signal obtained from the delay unit 127. The adder 123 outputs a signal obtained by combining the signal obtained from the adder 128 and the output signal of the filter 125.
That is, from the adder 123, the convolution operation unit 121
The signal obtained by synthesizing the signal obtained from the above, the signal obtained from the delay device 127, and the output signal of the filter 125 is output.

FIG. 9 is a diagram schematically showing the output signal of the adder 123. As described above, in the present embodiment, the value of the delay period T127 of the delay unit 127 corresponds to the time length (T1) of the initial reflected sound data to be convolved in the convolution operation unit 121. Therefore, in the impulse response of the reverberation adding unit 120, the adder 123
First outputs a signal (initial acoustic signal) obtained from the convolution operation unit 121, and subsequently outputs a signal (rear acoustic signal) obtained from the convolution operation unit 122. Here, the multiplication coefficients used in the convolution operation units 121 and 122 both use data based on the impulse response waveform of the same impulse response. Therefore, the connection between the signal (initial acoustic signal) obtained from the convolution operation unit 121 and the signal (rear acoustic signal) obtained from the convolution operation unit 122 becomes natural. There is no problem that the connection of data becomes unnatural in terms of hearing.

Further, the delay time T124 of the delay device 124
Is adjusted to be larger than the value of the delay period T127 of the delay unit 127, so that in the impulse response of the reverberation adding unit 120, the adder 123 uses the convolution operation unit 1
After outputting the signal (rear acoustic signal) obtained from 22,
The signal (reverberation signal) output from the filter 125 will be output. Here, the filter 125 generates a reverberation signal by repeatedly outputting while attenuating the signal obtained from the convolution operation unit 122. Therefore, the connection between the signal obtained from the convolution operation unit 122 (rear acoustic signal) and the signal output from the filter 125 (reverberation signal) becomes natural. There is no problem that the connection of data becomes unnatural in terms of hearing.

As described above, in the impulse response of the reverberation applying unit 120, the adder 123 is configured to operate in the convolution operation unit 1.
21 (initial acoustic signal), a signal obtained from the convolution operation unit 122 (rear acoustic signal), and a signal output from the filter 125 (reverberation signal) are output. Then, as schematically shown in FIG.
The output signal of the adder 123 is a signal which is well connected as data and which is natural to the sense of hearing.

Thereafter, the output signal (digital signal) of the adder 123 is supplied to the D / A conversion circuit 108 under the control of the CPU 105. The signal converted into an analog signal by the D / A conversion circuit 108 is output from the speaker 40 via the amplifier 30 as a sound with a reverberation effect.

Thus, the effector 100 according to the present invention
Then, in the sampling data of the impulse response waveform, an initial period from the time when the impulse sound is generated to the lapse of a predetermined time (for example, 0 seconds to
A signal related to the initial reflected sound (initial acoustic signal) is generated by convolving the convolution operation unit 121 with sampling data corresponding to a period of 0.3 seconds). Further, in the sampling data of the impulse response waveform, the second predetermined period after the initial period (for example, 0.3 seconds to 0.5 seconds).
The convolution operation unit 122 converts the sampling data corresponding to
A signal (rear acoustic signal) relating to the subsequent reflected sound is generated by convolving in.

Here, the reverberation applying unit 12 according to the present embodiment.
0 indicates that the operation result of the convolution operation unit 122 is the delay unit 12
7 and the delay device 124 (see FIG. 4). With such a configuration, the operation result of the convolution operation unit 122 is calculated by the adder 12
3 is output as it is as the reverberation addition result, while the attenuation characteristic is added in the filter 125. That is, the rear acoustic signal that is the calculation result of the convolution arithmetic unit 122 is used as a signal that follows the initial acoustic signal, and is also used as data for generating a reverberation signal. Therefore, for the acoustic space at the initial stage that characterizes the acoustic space, by using the convolution operation processing results of the convolution operation unit 121 and the convolution operation unit 122 that faithfully reproduce the data content of the impulse response waveform, sufficient acoustic characteristics can be obtained. Can be expressed. Further, regarding the acoustic space thereafter, the filter 12
The acoustic characteristics are expressed by using the signal obtained by performing the attenuation processing on the result of the convolution operation processing of the convolution operation unit 122 according to 5 as the reverberation signal. Therefore, since it is not necessary to convolve all the sampling data of the impulse response waveform, it is possible to express sufficient acoustic characteristics without requiring a huge hardware resource. Moreover, the reverberation signal is not generated based on the sampling data in the initial reflected sound period in which the reflected sound interval is relatively large, although the characteristic of the acoustic space appears, but rather in the rear part where the reflected sound interval becomes more precise. It is based on the sampling data of the reverberation period. Therefore, it is possible to generate a reverberation sound that more faithfully reproduces the acoustic characteristics of the impulse response wave system. Furthermore, since the signal related to the initial reflected sound, the signal related to the rear reverberation sound, and the signal related to the reverberation sound are all generated based on the sampling data of the same impulse response waveform, both signals are combined by the adder 123. At this time, there is no problem that the connection of signals becomes unnatural. That is,
It is possible to generate a signal in which the characteristics of the original reverberation space are added to the input signal. Further, by appropriately controlling the content of the reverberation characteristic added in the filter 125, it is possible to arbitrarily control the reverberation time while maintaining the data content of the impulse response waveform, in other words, while maintaining the characteristic of the acoustic space. it can.

B: Modified Example The embodiment of the present invention has been described above. However, this embodiment is merely an example and can be modified within the scope of the present invention. For example, the following may be considered as modifications.

(Modification 1) In the effector 100 according to the above-described embodiment, among the sampling data of the impulse response waveform, a predetermined period (0 seconds) has elapsed from the occurrence of the impulse sound (0 seconds).
The data included in the second reverberation sound is the data included in the first reflected sound data, and the data included in the predetermined period (0.3 to 0.5 seconds) is the rear reverberation sound data. However, this configuration can be changed arbitrarily.
For example, the data included in 0 to 0.2 seconds may be the initial reflected sound data, and the data included in the 0.2 to 0.5 seconds may be the rear reverberation sound data. That is, in the sampling data of the impulse response waveform, if the data for the initial reflected sound and the data for the rear reverberation sound are continuous, there is no problem that the connection of the data becomes unnatural to the sense of hearing.

(Modification 2) As shown in FIG. 10, a density adjusting filter 126 may be further provided after the filter 125. The density adjustment filter is a filter for adjusting (diffusing) the data density (pulse density) and the data phase (pulse phase) in the time axis direction. Here, the purpose of diffusing the data density in the time axis direction is that the sampling data of the general impulse response waveform enters the so-called diffusion region when a sufficient time has elapsed after the impulse sound was emitted, and the data (pulse signal) This is because it takes into consideration that the time interval at which the occurrence of is shortened. In addition, the purpose of spreading the phase of the data is to reproduce the acoustic characteristics in which the distinction of the phases disappears when a sufficient amount of time elapses, when the phase of the data of the impulse response waveform is considered as the balance of human left and right hearing. Is. Thus, by diffusing the data density and the data phase, it is possible to simulate the rear region (diffusion region) of the impulse response in the acoustic space.

Some specific examples of the structure of the density adjustment filter 126 will be listed below. A configuration in which all-pass filters (APF) are connected directly or in parallel. FIG. 11 shows z all-pass filters as 12APF-
1, 12APF-2, ..., 12APF-z are connected in series. FIG. 12 shows the all-pass filter 12
It is an example of an input waveform and an output waveform of the APF. As shown in FIG. 12, the all-pass filter 12APF-1
Is a filter having the function of inverting the phase of the first signal. Then, the all-pass filter is set to 12 APF-
By connecting in series with 1, 2, ..., Z, it is possible to generate and output signals whose phases are sequentially spread. In addition, as shown in FIG. 13, z number of all-pass filters 1
The 4APF and the multiplier 14A may be connected in parallel. Also in this case, it is possible to generate a signal whose phase is spread with respect to the input signal.

A configuration in which a feedback loop is formed using an all-pass filter (APF). FIG. 14 shows the output of the all-pass filter 15APF as a low-pass filter 15L, a delay device 15D and an amplifier 15.
This is a configuration in which feedback is provided via A. Figure 15
Shows an input waveform and an output waveform of the all-pass filter 15APF. All pass filter 15AP
F can spread the phase of the input signal. Furthermore, by adding the signals by feedback,
The time density of data can also be increased sequentially. Here, the fact that the time density of the data sequentially increases means that the intervals at which the data (pulse signals) exist in the time axis direction become successively shorter. When the time density of the data is sequentially increased in this manner, it is possible to obtain an effect that the pulses corresponding to the individual reflected sounds cannot be distinguished when judged by human hearing. Then, it becomes possible to reproduce the so-called diffusion area in the reverberation space.

A configuration using a multi-tap delay (FIG. 16). FIG. 17 illustrates an input waveform and an output waveform of the multi-tap delay 17MTD. The delay devices 17D-1, 2, ..., Q configuring the multi-tap delay 17MTD
If the delay times of are adjusted differently, the time density of data can be increased sequentially. Furthermore, the multiplier 17
The multiplication coefficient of A-0, 17A-1, ..., 17A-q is -1
It is also possible to spread the phase of data by setting the range of 1 to 1. Further, even with the configuration forming the feedback loop (FIG. 18), it is possible to similarly increase the time density of data and spread the phase of data.

Although a specific configuration example of the density adjustment filter 126 has been shown above, any one of these may be selected, and a plurality of density adjustment filters 12 may be combined.
6 may be configured.

(Modification 3) In the above-described embodiment, the sampling data of the impulse response waveform is actually measured. However, the sound field simulation program is stored in the ROM 102 in advance, and the user can make impulses in an arbitrary acoustic space. A simulated response waveform may be used as sampling data for convolution.

(Modification 4) Further, the user specifies the area range of the data used as "data for initial reflected sound" and "data for rear reverberation sound" of the sampling data of the impulse response waveform. It may be configured to be possible. With such a configuration, it is possible to select data that further characterizes the reverberation characteristics of the acoustic space.

(Modification 5) A plurality of impulse response waveform data may be stored in the reverberation data memory 107. In this case, as shown in FIG. 19, the names of acoustic spaces such as halls and churches corresponding to the respective impulse responses are also stored in the reverberation data memory 107 in association with each other. Then, the user may operate the operation unit 101 to select a desired acoustic space.

(Modification 6) Although the effector 100 equipped with the reverberation function has been described in the above embodiment, the present invention is also applicable to other devices such as mixers and reverbs equipped with the reverberation function. Is of course possible.

(Variation 7) A recording medium for recording the program according to the present invention is also optional, for example, a semiconductor memory,
CD-ROM (Compact Disc-Read Only Memory), C
Optical disc such as D-R (Compact Disc-Recordable), M
Examples include magneto-optical disks such as O (Magneto Optical Disc) and MD (Mini Disc), magnetic disks such as floppy (registered trademark) disks and hard disks. Also,
The method of installing such a program is also arbitrary, and may be installed in the effector 100 using the above-described recording medium, or may be installed in the effector 100 from a server storing the program according to the present invention via a network such as the Internet. Alternatively, a method using so-called online distribution may be used.

[0050]

As described above, according to the present invention,
Since the reverberation time can be arbitrarily controlled while maintaining the characteristics of the acoustic space, it is possible to impart a sufficient sound field effect with a simple configuration.

[Brief description of drawings]

FIG. 1 is a configuration diagram of an effector 100 according to an embodiment of the present invention.

FIG. 2 is a diagram schematically showing sampling data of impulse response waveforms.

FIG. 3 is a diagram schematically showing data contents of a reverberation data memory 107 of the effector 100 according to the embodiment of the present invention.

FIG. 4 is a configuration diagram of a reverberation imparting unit 120 of the same effector 100.

FIG. 5 is a convolution operation unit 121 of the effector 100.
It is a block diagram of.

FIG. 6 is a convolution operation unit 122 of the same effector 100.
It is a block diagram of.

FIG. 7 is a diagram for explaining the signal processing content of a filter 125 of the same effector 100.

FIG. 8 is a diagram for explaining the signal processing content of a filter 125 of the same effector 100.

FIG. 9 is a diagram for explaining the signal processing content of the reverberation imparting unit 120 of the same effector 100.

FIG. 10 is a configuration diagram of an effector according to a second modified example of the present invention.

FIG. 11 is a configuration example of a density adjustment filter 126 for an effector according to a second modified example of the present invention.

FIG. 12 is a diagram for explaining the signal processing content of the same density adjustment filter 126.

FIG. 13 is another configuration example of the same density adjustment filter 126.

FIG. 14 is another configuration example of the same density adjustment filter 126.

FIG. 15 is a diagram for explaining the signal processing content of the same density adjustment filter 126.

FIG. 16 is another configuration example of the same density adjustment filter 126.

FIG. 17 is a diagram for explaining the signal processing content of the same density adjustment filter 126.

FIG. 18 is another configuration example of the same density adjustment filter 126.

FIG. 19 is a diagram for explaining a fifth modified example of the present invention.

[Explanation of symbols]

100 ... effector, 101 ... operation part, 102 ... ROM (Read Only Memory), 103 ... RAM (Random Access Memory), 104 ... A / D (Analog / Digital) conversion circuit, 105 ... CPU (Central Processing Unit), 106 ... Display, 107 ... Reverberation data memory, 108 ... D / A (Digital / Analog) conversion circuit, 109 ... bus, 120 ... Reverberation unit, 10 ... microphone, 30 ... amplifier, 40 ... Speaker.

Continued front page    (72) Inventor Kenichi Tamiya             Yamaha stock, 10-1 Nakazawa-machi, Hamamatsu-shi, Shizuoka             In the company (72) Inventor Satoshi Sekine             Yamaha stock, 10-1 Nakazawa-machi, Hamamatsu-shi, Shizuoka             In the company F-term (reference) 5D108 AB05 AB07 AB19 AD04 AD05                       AD07 AD08

Claims (9)

[Claims] 1. An initial acoustic signal is generated by convolving sampling data corresponding to an initial period from the time when an impulse sound is generated in an impulse response waveform to the time when a first predetermined time has elapsed with respect to an acoustic signal to be processed. 1
A convolution calculation unit for generating a rear acoustic signal by convolving the acoustic signal with sampling data corresponding to a period from the lapse of the initial period of the impulse response waveform to the lapse of a second predetermined time. A convolution operation unit, an attenuation operation unit that repeatedly outputs the rear acoustic signal while attenuating the rear acoustic signal, and an output unit that combines and outputs the initial acoustic signal, the rear acoustic signal, and the reverberation signal. A reverberation imparting device comprising: 2. An initial acoustic signal is generated by convolving sampling data corresponding to an initial period from the time when an impulse sound is generated in an impulse response waveform until the first predetermined time has elapsed with respect to the acoustic signal to be processed. 1
A convolution calculation unit for generating a rear acoustic signal by convolving the acoustic signal with sampling data corresponding to a period from the lapse of the initial period of the impulse response waveform to the lapse of a second predetermined time. And an attenuation calculator that repeatedly outputs while attenuating the rear acoustic signal to generate an attenuation signal, and a diffusion that generates a reverberation signal by adjusting at least one of density and phase of the attenuation signal. A reverberation imparting apparatus, comprising: a unit; and an output unit that synthesizes and outputs the initial acoustic signal, the rear acoustic signal, and the reverberation signal. 3. The reverberation applying apparatus according to claim 1, further comprising a storage unit that stores sampling data to be convoluted in the first convolution operation unit and the second convolution operation unit. 4. The output unit outputs the rear acoustic signal by adding a delay corresponding to a time length of sampling data corresponding to the initial period. The reverberation applying device described in. 5. An initial acoustic signal is generated by convolving sampling data corresponding to an initial period from the time when an impulse sound is generated in the impulse response waveform until the first predetermined time has elapsed with respect to the acoustic signal to be processed. A convolution calculation step of 1, and generating a rear acoustic signal by convolving the acoustic signal with sampling data corresponding to a period from the lapse of the initial period of the impulse response waveform to the lapse of a second predetermined time. 2, a convolution calculation process for generating a reverberation signal by repeatedly outputting the rear acoustic signal while attenuating the rear acoustic signal, and an output for combining the initial acoustic signal, the rear acoustic signal, and the reverberation signal to output. A reverberation imparting method comprising: 6. An initial acoustic signal is generated by convolving sampling data corresponding to an initial period from the time when an impulse sound is generated in the impulse response waveform until the first predetermined time has elapsed with respect to the acoustic signal to be processed. A convolution calculation step of 1, and generating a rear acoustic signal by convolving the acoustic signal with sampling data corresponding to a period from the lapse of the initial period of the impulse response waveform to the lapse of a second predetermined time. 2, a convolution calculation step, an attenuation calculation step of repeatedly outputting while attenuating the rear acoustic signal to generate an attenuation signal, and at least one of a density and a phase of the attenuation signal is adjusted to generate a reverberation signal. The diffusion process, the initial acoustic signal, the rear acoustic signal, and the reverberation signal are combined and output. Reverberation applying method characterized by comprising an output process, the. 7. The initial acoustic signal is generated by causing a computer to convolve the sampling data corresponding to the initial period from the time when the impulse sound is generated in the impulse response waveform until the first predetermined time elapses with the acoustic signal to be processed. A first convolution calculation means for generating a rear acoustic signal by convolving the acoustic signal with sampling data corresponding to a period from the passage of the initial period of the impulse response waveform to the passage of a second predetermined time. Second convolution calculation means for generating; attenuation calculation means for generating a reverberation signal by repeatedly outputting while attenuating the rear acoustic signal; and combining the initial acoustic signal, the rear acoustic signal and the reverberation signal. A program that functions as output means for outputting. 8. The initial acoustic signal is obtained by convolving a computer with sampling data corresponding to an initial period from when an impulse sound is generated in an impulse response waveform to after a first predetermined time elapses with respect to the acoustic signal to be processed. A first convolution calculation means for generating a rear acoustic signal by convolving the acoustic signal with sampling data corresponding to a period from the passage of the initial period of the impulse response waveform to the passage of a second predetermined time. Second convolution calculation means for generating; attenuation calculation means for generating an attenuation signal by repeatedly outputting the rear acoustic signal while attenuating the rear acoustic signal; and reverberation signal by adjusting at least one of density and phase of the attenuation signal. And a diffusing means for generating, combining the initial acoustic signal, the rear acoustic signal, and the reverberation signal. Program for functioning as an output unit for force. 9. A computer-readable recording medium in which the program according to claim 7 or 8 is recorded.