JP2003263179A - 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: A first reflected sound signal and a second reflected sound signal following the first reflected sound signal are generated by convoluting first and second data strings corresponding to the prescribed time of an impulse response into an acoustic signal to be processed. The two signals are combined so as to be overlapped to each other. Preferably, fade-out processing is applied to the first reflected sound signal, fade-in processing is applied to the second reflected sound signal and the two signals are combined so as to be cross-faded with each other. <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 reverberation imparting device for imparting reverberation effects in various acoustic spaces to an acoustic signal.

[0002]

2. Description of the Related Art By measuring an impulse response waveform in an acoustic space such as a hall or a church and convolving sampling data of the impulse response waveform into an acoustic signal, an effect of an early reflection sound and a reverberation sound after that in the acoustic space can be obtained. There is provided a device for imparting sound (hereinafter referred to as reverberation imparting device). The sampling data of the impulse response waveform in the acoustic space is used for impulse sound or TSP (Time Stret) generated from a sound source installed in the acoustic space.
It can be obtained by picking up an acoustic measurement signal such as a ched pulse) with a microphone or the like, then 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 some time has passed since the generation of the impulse sound, so that the accurate sound In order to reproduce the reverberation space, the impulse response waveform takes a long time. Then, in order to reproduce such a reverberant space, enormous hardware resources for convolving a large amount of sampling data are required.

Therefore, conventionally, convolution calculation processing using all sampling data of the impulse response waveform is not performed, but convolution using only data included in the impulse response waveform data within a predetermined period from the generation of the impulse sound. The data related to the initial reflected sound was generated by performing the calculation. Subsequent reflected sound after that (reverberation sound)
The data related to (1) was separately added artificially, and the reverberation characteristics were added. However, in this method, since the data of the initial reflected sound and the data of the subsequent reflected sound (reverberation sound) are different in sound quality, when the data of the reverberation sound is followed by the data of the initial reflected sound, The connection of data in the connection part was bad, which was not preferable in terms of hearing. With the conventional method, although the scale of the hardware resource can be suppressed to a small level, it has not been possible to express a sufficient sound field effect.

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

[0006]

In order to solve the above-mentioned problems, the reverberation imparting apparatus according to the present invention has a first reverberation corresponding to a first predetermined time of an impulse response to an acoustic signal to be processed. First signal processing means for generating a first reflected sound signal by convolving the data string, and second reflected sound that follows the first reflected sound signal by subjecting the acoustic signal to predetermined signal processing. Second signal processing means for generating a signal, the first reflected sound signal generated by the first signal processing means, and the second reflected sound signal generated by the second signal processing means And a means for adding and outputting the second reflected sound signal, the output means delaying the second reflected sound signal by a time shorter than the first predetermined time and outputting the delayed second sound signal. According to the configuration of the reverberation applying apparatus, the first reflected sound signal can be generated by convolving only the first data string corresponding to the predetermined time of the impulse response data. In addition, the acoustic signal to be processed is subjected to a predetermined signal processing to perform the first signal processing.
A second reflected sound signal subsequent to the reflected sound signal can be generated. Then, the second part is formed so that the end part of the first reflected sound signal and the start part of the second reflected sound signal overlap each other.
It is possible to output the signal obtained by delaying the reflected sound signal of (1) and then performing the synthesis processing as a signal with reverberation. For this reason, the reverberation effect can be added without convolving all the impulse response data with respect to the acoustic signal to be processed, and the output signal subjected to the reverberation processing is related to the first reflected sound signal. There is no sudden change from the sound quality to the sound quality related to the second reflected sound signal,
No audible unnatural situation occurs.

Here, the second signal processing means outputs the second reflected sound signal by repeatedly outputting the first reflected sound signal generated by the first signal processing means using a recursive filter. It may be characterized in that it is a means for generating. According to this configuration, the second reflected sound signal can be generated by repeatedly outputting the first reflected sound signal generated by the first signal processing means using the cyclic filter. For this reason, it is not necessary to convolve all the impulse response data, and the second reflected sound signal can be generated with a simple configuration.

The reverberation imparting apparatus according to the present invention generates the first reflected sound signal by convolving the first data sequence corresponding to the first predetermined time of the impulse response with the acoustic signal to be processed. A first signal processing means and a second data string corresponding to a second predetermined time of the impulse response are convoluted with the acoustic signal, and the convoluted signal is repeatedly output using a cyclic filter. By doing so, second signal processing means for generating a second reflected sound signal following the first reflected sound signal, and the first reflected sound signal generated by the first signal processing means, A means for adding and outputting the second reflected sound signal generated by the second signal processing means, wherein the second reflected sound signal is output for a time shorter than the first predetermined time. Delay It may be characterized by comprising output means for force, the. According to the configuration of the reverberation imparting apparatus, it is possible to generate the first reflected sound signal by convolving the first data string corresponding to the first predetermined time of the impulse response with the acoustic signal to be processed. it can.
Further, the second reflected sound signal can be generated by convolving the second data string corresponding to the second predetermined time with the acoustic signal to be processed. Then, the second reflected sound signal is delayed so that the end portion of the first reflected sound signal and the start portion of the second reflected sound signal overlap each other, and the synthesis processing is performed, and reverberation is added. It can be output as a signal. Therefore, it is possible to add a reverberation effect without the need to convolve all impulse response data with respect to the acoustic signal to be processed, and
In the reverberation-added output signal, the sound quality of the first reflected sound signal does not suddenly change to the sound quality of the second reflected sound signal, and an unnatural audible situation does not occur.

Further, the reverberation applying apparatus according to the present invention uses a signal obtained by convolving a sound signal to be processed with a first data string corresponding to a first predetermined time of an impulse response to the impulse signal. First signal processing means for generating a first reflected sound signal by following the signal obtained by convolving the second data string corresponding to the second predetermined time of the response; and the first signal. In the processing means, a signal obtained by convolving the second data string is repeatedly output using a recursive filter to generate a second reflected sound signal that follows the first reflected sound signal. Two signal processing means, the first reflected sound signal generated by the first signal processing means, and the second reflected sound signal generated by the second signal processing means. And means for outputting the calculated, the second reflected sound signals, the said first predetermined time second
The output means may delay the output by a time shorter than a value obtained by adding the predetermined time of, and output.

In outputting the composite signal of the first reflected sound signal and the second reflected sound signal, the first reflected sound signal and the second reflected sound signal are cross-faded and output. It is preferable that According to such an aspect, in the reverberation-added output signal, the first signal
The sound quality related to the reflected sound signal changes naturally according to the sound quality related to the second reflected sound signal, which makes the sound more natural.

Further, in outputting a composite signal of the first reflected sound signal and the second reflected sound signal, the signal level of either or both of the first reflected sound signal and the second reflected sound signal is adjusted. It is also possible to have a means for doing so.
According to this aspect, in the reverberation-added output signal, the sound quality and volume of the second reflected sound signal naturally change from the sound quality of the first reflected sound signal. It will be natural.

In adjusting the signal level, the level value of the signal included in the first predetermined time of the first reflected sound signal and the level value of the signal included in the predetermined time of the second reflected sound signal are set. It is preferable to adjust the level accordingly.

In the reverberation applying method according to the present invention, the first reflected sound signal is generated by convolving the first data string corresponding to the predetermined time of the impulse response with the acoustic signal to be processed. A signal processing step, a second signal processing step of performing a predetermined signal processing on the acoustic signal to generate a second reflected sound signal following the first reflected sound signal, and the first signal processing A step of adding and outputting the first reflected sound signal generated in the step and the second reflected sound signal generated in the second signal processing step, the second reflected sound signal And outputting the signal after delaying the signal by a time shorter than the predetermined time.

In the reverberation applying method according to the present invention, the first reflected sound signal is generated by convolving the first data sequence corresponding to the first predetermined time of the impulse response with the acoustic signal to be processed. A first signal processing step of generating and a second data string corresponding to a second predetermined time of the impulse response are convoluted with the acoustic signal, and the convoluted signal is generated using a recursive filter. By repeatedly outputting, the second reflected sound signal following the first reflected sound signal is output.
A second signal processing step of generating a reflected sound signal of the above, the first reflected sound signal generated by the first signal processing step, and the second signal generated by the second signal processing step of A step of adding the reflected sound signal and outputting the delayed reflected sound signal, the step of outputting the second reflected sound signal after delaying the second reflected sound signal by a time shorter than the first predetermined time. May be

A program according to the present invention generates a first reflected sound signal by causing a computer to convolve a first data string corresponding to a first predetermined time of an impulse response with an acoustic signal to be processed. A second signal processing means for performing a predetermined signal processing on the acoustic signal to generate a second reflected sound signal subsequent to the first reflected sound signal; Means for adding and outputting the first reflected sound signal generated by the first signal processing means and the second reflected sound signal generated by the second signal processing means. It is a program for causing the reflected sound signal No. 2 to function as an output unit for delaying and outputting the reflected sound signal by a time shorter than the first predetermined time.

Also, the program according to the present invention causes the computer to convolve a first data string corresponding to a first predetermined time of an impulse response with respect to an acoustic signal to be processed, thereby producing a first reflected sound signal. And a second signal sequence corresponding to a second predetermined time of the impulse response is convoluted with the acoustic signal, and the convoluted filter is used to generate the convoluted signal. Second signal processing means for generating a second reflected sound signal following the first reflected sound signal by repeatedly outputting the first reflected sound signal, and the first reflected sound generated by the first signal processing means. A means for adding and outputting a signal and the second reflected sound signal generated by the second signal processing means, wherein the second reflected sound signal is output for a time longer than the first predetermined time. Short It may be one which is a program for functioning as an output means for outputting delayed by time.

[0017]

BEST MODE FOR CARRYING OUT THE INVENTION Next, preferred embodiments of the present invention will be described with reference to the drawings.

A: First Embodiment A1: Configuration FIG. 1 is a reverberation applying apparatus 10 according to an embodiment of the present invention.
It is a block diagram of 0. As shown in FIG. 1, the reverberation applying device 1
00 includes an operation unit 1, a controller 2, an input unit 3, an output unit 4, a DSP 5, and a sound source 6, and each unit is connected via a bus 9.

A microphone MIC is connected to the input unit 3, and the user inputs an acoustic signal to be reverberation added,
It is supplied to the input unit 3 via the microphone MIC. The input unit 3 outputs to the controller 2 the supplied acoustic signal that is A / D (digital / analog) converted at a predetermined sampling period. The sound source 6 has a function of generating a predetermined acoustic signal, and the user can also add a reverberation effect to the acoustic signal output from the sound source 6.

A speaker SP is connected to the output unit 4. Under the control of the controller 2, the output unit 4 D / A (digital / analog) converts the reverberation-added digital signal and outputs it to the speaker SP. Speaker S
P outputs the acoustic signal (analog signal) that has been subjected to the reverberation processing to the outside.

Operation keys and a liquid crystal panel are arranged in the operation unit 1, and the user operates the operation keys of the operation unit 1 to instruct a specific reverberation adding process. The operation unit 1 outputs the contents of the operation keys operated by the user to the controller 2.
The controller 2 is a CPU (Central Processing Uni
t) and is provided with the reverberation applying device 10 based on a user's instruction.
It controls each unit of 0 to execute various processes related to reverberation. Further, the controller 2 has a memory, and a control program relating to reverberation is stored in advance in the memory.

Under the control of the controller 2, the DSP 5 executes a specific reverberation adding process. Figure 2 shows DS
It is a figure which shows the structure of P5, and thus DSP5 is comprised from the reverberation data memory 107 and the reverberation addition part 120.

The reverberation data memory 107 stores sampling data of impulse response waveforms in various acoustic spaces. The sampling data of the impulse response waveform is obtained in advance by actual measurement and simulation. FIG. 3 schematically shows an example of sampling data of the impulse response waveform. In FIG. 3, the horizontal axis represents time, the vertical axis represents signal level, and the sampling time is TS. Here, the reverberation data memory 107 according to the present embodiment does not store all the sampling data of the impulse response waveform, but a part of the sampling data. More specifically, the data string D1 included in a period T1 of a predetermined period (for example, 0 to 0.2 seconds) from the time when the impulse sound is generated (0 second) and another predetermined period (for example, 0.3 to 0). .6
The data string D2 included in the period T2 of (second) is stored in the reverberation data memory 107. As illustrated in FIG.
The data sequence D1 and the data sequence D2 have a discontinuous relationship with each other in the original impulse response waveform.

More specifically, the data strings D1 and D2 are not stored in the reverberation data memory 107 as they are, but a predetermined data correction is performed on each of the data strings D1 and D2. The reverberation data (referred to as a data string D1 ′ and a data string D2 ′) are stored in the reverberation data memory 107. The reason why the data strings D1 and D2 are corrected is to make the signal to which the reverberation effect has been applied more natural to the sense of hearing. The details of this specific data correction will be described later.

Returning to FIG. 2 again, the configuration of the DSP 5 will be described. The reverberation imparting unit 120 specifically performs reverberation effect imparting processing on the supplied acoustic signal (sampling data). FIG. 4 is a block diagram of the reverberation imparting unit 120. As described above, the reverberation imparting unit 120 includes the convolution operation unit 121,
Convolution operation unit 122, combining unit 123, delay device 124,
The filter 125 is provided.

The convolution operation unit 121 applies the reverberation data memory 1 to the sampling data of the supplied acoustic signal.
The arithmetic processing for convolving the data string D1 ′ stored in 07 is performed. The convolution operation unit 121 outputs a signal obtained by convolving the data string D1 ′ as an initial reflected sound signal S121. As shown in FIG. 5, the convolution operation unit 121 includes delay units 121D-1, 121D-2, ..., 121D- (m
-1), the multipliers 121A-0, 121A-1, 121A
--2, ..., 121A- (m-1), adder 121K-
1, 121K-2, ..., 121K- (m-1), and performs m stages of convolution operation processing. Delay devices 121D-1, 121D-2, ..., 121D- (m-
Each delay time of 1) corresponds to the sampling time Ts of the impulse response waveform. Here, the total sum of the delay times of the delay devices is T121. Also, the multipliers 121A-0, 1
21A-1, 121A-2, ..., 121A- (m-
As each multiplication coefficient of 1), the data string D1 ′ for initial reflected sound (La1, La2, ...) In the reverberation data memory 107.
To set. Specifically, as the multiplication coefficient of the multiplier 121A-0, the initial reflected sound data La1 and the multiplier 121A-
As the multiplication coefficient of 1, the initial reflected sound data La2, ...,
Thus, the initial reflected sound data Lam is set as the multiplication coefficient of the multiplier 121A- (m-1).

The convolution operation unit 122 performs an operation process of convoluting the data sequence D2 'stored in the reverberation data memory 107 with respect to the sampling data of the acoustic signal. The convolution operation unit 122 outputs the signal obtained by convolving the data string D2 ′ as the rear reflected sound signal S122. As shown in FIG. 6, the convolution operation unit 122 includes a delay unit 122D-
1, 122D-2, ..., 122D- (n-1), multipliers 122A-0, 122A-1, 122A-2 ,.
122A- (n-1), adders 122K-1, 122K
, ..., 122K- (n-1), and performs n-stage convolution operation processing. Delay device 122D-
Each of the delay times of 1, 122D-2, ..., 122D- (n-1) is the sampling time Ts of the impulse response waveform.
Corresponding to. Here, the sum of the delay times of the delay units is T1.
22. Further, the multipliers 122A-0 and 122A-
2, ..., 122A- (n-1) as multiplication coefficients,
Rear reverberation sound data string D2 in the reverberation data memory 107
'(Lb1, Lb2, ...) Is set. In particular,
The rear reverberation sound data Lb1 is set as the multiplication coefficient of the multiplier 122A-0, the rear reverberation sound data Lb2 is set as the multiplication coefficient of the multiplier 122A-1, and the like, and the multiplier 122A- (n-1) is set. The rear reverberation sound data Lbn is set as the multiplication coefficient of

Returning to FIG. 4, the structure of the reverberation unit 120 will be described. The delay device 124 delays the rear reflection signal S122 supplied from the convolution operation unit 122 by a predetermined time T124 and outputs the delayed signal to the filter 125. Here, in the conventional reverberation imparting apparatus, a synthesis process is performed in which the initial reflected sound signal S121 output from the convolution operation unit 121 is simply followed by the rear reflected sound signal S122 output from the convolution operation unit 122. Therefore, delay device 1
The delay time T124 of 24 corresponds to the time length T121, that is, the time length of the data string D1 (data string D1 ′) (time corresponding to the period T1 in FIG. 3 above: 0.2 seconds). It was In this embodiment, unlike the conventional case, the delay time T124 of the delay device 124 is the time length T
A value that is shorter than 121 by a predetermined time α (T124 = T
121-α) is characteristic. The meaning of the value of the delay time T124 will be described later.

The filter 125 is a recursive filter having a feedback loop, and in this embodiment, a comb filter (Comb filter) as shown in FIG. 5 is adopted.
More specifically, as shown in FIG. 4, the filter 125 includes a filter 125F including delay devices 125D and 125ID, a low pass filter 125L, amplifiers 125A and 125GA, and an adder 125K.
F-1, 125F-2, ..., 125F-p and p stages connected in parallel. Here, the delay device 125ID is
It plays a role as an initial delay that gives a predetermined delay to the input signal of the filter 125F-1. Further, the amplifier 125GA adjusts the level of the entire output signal of the filter 125F-1. It should be noted that the low-pass filter may be any one that attenuates high frequencies, and a shelving filter may be used. FIG. 7 shows the filter 125F-1.
It is the figure which showed typically the output signal when one pulse signal is supplied to. As shown in FIG. 7, the filter 12
The output signal of 5F-1 is a signal in which data is followed at each delay time T125 by the delay device 125D,
If the amplification coefficient of the amplifier 125A is adjusted to a number less than 1, it is possible to output a signal having a characteristic of repeatedly attenuating from the filter 125F-1 as shown in FIG. Here, the low-pass filter 125L forming the filter 125F-1 has a function of removing a high frequency signal component having a predetermined frequency or higher. Therefore, the filter 1 of each stage
25F-1, 125F-2, ..., 125F-p, by adjusting the filter coefficient of the low-pass filter and the amplification coefficient of the amplifier that constitutes the filter of each stage, for example, the reverberation time becomes shorter for higher frequency signals. It is possible to reproduce the acoustic characteristic of becoming.

FIG. 8 shows an output signal S122 when a pulse signal is supplied to the convolution operation unit 122 and each filter 1
25F-1, 125F-2, ..., 125F-p output signals S125-1, S125-2 ,.
125-p and the output signal S125 of the filter 125 are shown schematically. Convolution operation unit 12
The signal S122 obtained from 2 is obtained by each filter 125F-
1, 125F-2, ..., 125F-p, and then each filter outputs a signal component corresponding to the frequency band of the filter while repeatedly attenuating. Then, the output signals S125-1, S125- of each filter
The combined signal S125 of 2, ..., S125-p is output from the filter 125. As a result, as shown in FIG. 8, as compared with the time length (period) T122 of the signal S122 obtained from the convolution operation unit 122, the filter 125 generates and outputs the attenuated signal S125 having a longer period (T125> T122). Can be made. Hereinafter, this attenuation signal S125 is described as a reverberation sound signal S125. Further, for convenience of description, in FIG. 8, the output signals S125-1, S125-2, ..., S125 of the filters 125F-1, 125F-2 ,.
-P has the same signal shape, but actually each filter 125F-1, 125F-2, ..., 125F-
Depending on the value of the filter coefficient of p, the output signal S125-1,
The signal shape of S125-2, ..., S125-p (attenuating slope, period, etc.) is different.

Returning to FIG. 4, each section of the reverberation adding section 120 will be described. The synthesizing unit 123 includes the initial reflected sound signal S121 output from the convolution operation unit 121 and the filter 125.
Signal S1 obtained by adding the reverberation sound signal S125 output from
Is a means for outputting 23. This signal S123 is DS
From P5, it is output as the reverberation adding result. Figure 9
Is the initial reflected sound signal S12 supplied to the synthesis unit 123.
1 schematically shows the time relationship of the reverberation sound signal S125. Here, for convenience of description, the contents of the signal S123 are also shown. As described above, the delay unit 124
Lags the rear reflected sound signal S122 output from the convolution operation unit 122 by a time T124, and then the filter 125 generates a reverberant sound signal S125 based on the rear reflected sound signal S122. Therefore, the filter 12
Assuming that the time required for signal processing in 5 is negligible compared to the delay time T124, as shown in FIG. 9, only the time T124 has elapsed since the initial reflected sound signal S121 was supplied to the synthesizing unit 123. After that, the reverberation sound signal S125 starts to be supplied. Since the delay time T124 is a value shorter than the time length T121 of the initial reflected sound signal S121 (T124 = T121-α), a period Tα in which the initial reflected sound signal S121 and the reverberation sound signal S125 overlap with each other always occurs. It will be.

As shown in FIG. 9, the combined signal S123 generated by the combining section 123 is composed of three parts, that is, a signal S123-1, a signal S123-2 and a signal S123-3. Of these, the signal S123-1 is a signal based on only the initial reflected sound signal S121, and
123-3 is a signal based only on the reverberation sound signal S125. The signal S123-2 is the initial reflected sound signal S.
It is a combined signal in which both 121 and the reverberation sound signal S125 are mixed. As described above, the signal S123 output from the synthesizing unit 123 has the signal S123-2 in which the initial reflected sound signal S121 and the reverberant sound signal S125 are mixed, so that the signal property (sound quality) is the initial reflected sound. There is no sudden change from the signal S121 to that of the reverberation sound signal S125. Therefore, the signal S1 output from the DSP 5
Then, even when 23 is output from the speaker SP via the output unit 4, the sound quality suddenly changes as in the conventional case, and the situation where the sound is unnatural does not occur. More specifically, the sound quality based on the initial reflected sound signal S121 gradually changes to the sound quality based on the reverberation sound signal S125. Particularly, in the present embodiment, since the data sequence D1 that is convolved in the convolution operation unit 121 and the data sequence D2 that is convolved in the convolution operation unit 122 are not in a continuous relationship in the original impulse response data, the initial reflection There is a high possibility that the sound quality of the sound signal S121 and the sound quality of the reverberation sound signal S125 generated from the rear reflected sound signal S122 are different. However, even in such a case, the reverberation imparting apparatus 100 according to the present embodiment does not cause a sudden change in sound quality as in the conventional case, and performs the synthesis process so that the sound quality gradually changes. Therefore, the reverberation characteristic which is natural in hearing can be given. The relationship in the original impulse response of the data string D1 and the data string D2 is arbitrary, and for example, a temporally continuous relationship, a temporally overlapping relationship, and a temporally completely overlapping relationship ( That is, they may have the same relationship).

The above is the reverberation applying apparatus 1 according to the present embodiment.
00 is an outline. As described above, the greatest feature of the reverberation imparting apparatus 100 according to this embodiment is that the initial reflected sound signal S
When the 121 and the reverberation sound signal S125 are combined, the reverberation sound signal S1 is added after the initial reflection sound signal S121 as in the conventional case.
25 is not directly followed, but a point where the initial reflected sound signal S121 and the reverberant sound signal S125 are overlapped with each other is provided at the connection portion of the initial reflected sound signal S121 and the reverberant sound signal S125.

By the way, in the reverberation imparting apparatus 100 according to the present embodiment, in performing the processing in which the reverberation sound signal S125 follows the initial reflected sound signal S121, the impulse is changed so that the sound quality of the auditory sense becomes more natural. Correction is performed on the data strings D1 and D2 of the response waveform. The details of the correction for the data strings D1 and D2 will be described below.

A2: Data string D of impulse response waveform
Correction for 1 and D2 FIG. 10 shows the data sequence D1 in the impulse response waveform.
(Dashed line) and the contents of the data string D1 ′ after data correction are schematically shown. The time axis shows the start time of the data string D1 (D1 ') as the reference time (0 seconds). As shown in FIG. 10, the correction on the data string D1 is performed on the data portion included in the predetermined time (time α in the present embodiment) immediately before the end of the data string D1. Specifically, for the data value immediately before the end of the data string D1, the data value gradually decreases,
That is, as the data value fades out,
It is corrected (fade-out correction). Here, the fade-out mode of the data string D1 ′ is arbitrary, for example, 1
The data value may be faded out so as to monotonically decrease along the value given by the next function, or may be along the value given by another function such as a logarithmic function. According to a simulation experiment conducted by the applicant, y = log
When a logarithmic function of (a × x + b) (x is the value of the data string D1, y is the value of the data string D1 ′, and a and b are predetermined constants) is used, it is possible to achieve a reverberation effect that is preferable for hearing. all right.

FIG. 11 schematically shows the contents of the data string D2 (broken line) in the impulse response waveform and the data string D2 'after data correction. In the time axis, the start time of the data string D2 (D2 ') is the reference time (0
Seconds). As shown in FIG. 11, the correction for the data string D2 is performed on the data portion included in the predetermined time (time α in the present embodiment) immediately after the start of the data string D2. Specifically, the data value immediately after the start of the data string D2 is corrected (fade-in correction) so that the data value gradually increases from 0, that is, the data value fades in. It Here, the fade-in mode of the data string D2 ′ is arbitrary, and may be fade-in so that the data value monotonically increases along the value given by the linear function, and is given by another function. You may make it follow a value.

FIG. 12 schematically shows specific data contents stored in the reverberation data memory 107.
FIG. 12 shows time information when the impulse sound is generated and a data string D1 ′ (data La1, La2, ...).
..., Lam) and the data string D2 '(data Lb1, L)
b2, ..., Lbn) are stored in association with each other, but the sampling time T is omitted by omitting the time information.
The values of the data string D1 ′ and the data string D2 ′ may be stored as the time series data for each S. Here, as the values of the data string D1 ′ and the data string D2 ′, values corresponding to the sampling data values of the impulse response waveform may be stored, and values standardized at a certain level of the sampling data of the impulse response waveform are stored. You may do it.

In this way, the reverberation data memory 107
The data string D1 ′ stored in the
In, the process of convoluting the acoustic signal to be processed is performed, and the initial reflected sound signal S121 is generated. Figure 1
3 is a diagram schematically showing the initial reflected sound signal S121. In this way, since the fade-out processed data sequence D1 ′ is convoluted in α time before the end of the convolution operation, the initial reflected sound signal S121 is convoluted. The α-time portion before the end of the signal S121 is also fade-out processed.
In addition, the data string D stored in the reverberation data memory 107
In 2 ′, the convolution operation unit 122 performs a convolution process on the acoustic signal to be processed, and a rear reflected sound signal S122 is generated. FIG. 14 shows the rear reflected sound signal S.
FIG. 122 is a diagram schematically showing 122. However, since the fade-in processed data sequence D2 ′ is convoluted in α time after the start of the convolution operation in this manner, after the rear reflected sound signal S122 is started. The α time portion is also fade-in processed.

The rear reflected sound signal S122 is then repeatedly attenuated by the filter 125 and output as a reverberant sound signal S125. FIG. 15 shows the reverberation sound signal S
It is the figure which showed the content of 125 typically, However, the rear part reflected sound signal S122 itself supplied to the filter 125 from the convolution operation part 122 has become the form in which the fade-in process was carried out, and the reverberation sound signal S125 also fades in. It has been processed.

As described above, the convolution operation unit 121
Initial reflected sound signal S12 that has been faded out from
1. The reverberation sound signal S125 that has been subjected to the fade-in process is output from the filter 125, and these two signals S12
1, S125 are supplied to the combining unit 123. FIG.
Is the initial reflected sound signal S12 supplied to the synthesis unit 123.
1 shows the reverberation sound signal S125 schematically.
As described above, the initial reflection sound signal S12 is sent to the synthesis unit 123.
1 is started to be supplied, the delay time T1 of the delay device 124 is
After a delay of 24, the reverberation sound signal S125 begins to be supplied. The delay time T124 of the delay device 124 is smaller than the time length T121 of the initial reflected sound signal S121 by a predetermined time (α time) (T124 = T1).
21-α). That is, as shown in FIG. 16, the fade-out processed portion of the initial reflected sound signal S121 (α time immediately before the end of the initial reflected sound signal S121) and the fade-in processed portion of the reverberation sound signal S125 (reverberation The sound signal S125 is just overlapped with α time immediately after the start). This overlapping period is shown as a period Tα (see FIG. 16).

As described above, according to the reverberation imparting apparatus 100 of this embodiment, the initial reflected sound signal S relating to the initial reflected sound is obtained.
When 121 and the reverberation sound signal S125 related to the rear reflection sound are combined, the initial reflection sound signal S121 and the reverberation sound S121 are not directly followed by the reverberation sound signal S125 as in the conventional case. An initial reflected sound signal S121 and a reverberation sound signal S12 are provided at a connection portion of the signal S125.
A period Tα for overlapping 5 is provided. Further, in the overlap period Tα, the initial reflected sound signal S121 is faded out, and the reverberation sound signal S125 is generated.
Is in a state in which the fade-in processing has been performed. As a result, the initial reflected sound signal S121 and the reverberation sound signal S125 are cross-fade synthesized, and the synthesized signal S12
In No. 3, the effect that the change of the sound quality becomes more natural and smooth is obtained.

By the way, in performing the processing in which the reverberation sound signal S125 follows the initial reflected sound signal S121, it is actually necessary to consider not only the sound quality but also the level (volume) of the acoustic signal. As shown in FIG. 16, when there is no great difference between the signal levels of the initial reflected sound signal S121 and the reverberation sound signal S125, the level (volume) of the acoustic signal changes even when these two signals are combined. It will not be unnatural. However, the data sequence D1 and the data sequence D2 in the original impulse response waveform data are
When the time positions of are largely separated, as shown in FIG. 17, the initial reflected sound signal S121 and the reverberation sound signal S1 are
It is also envisaged that there will be a large difference in the overall 25 signal levels. In this case, in the synthesized signal S123, the signal level (volume) changes unnaturally as shown in FIG. 18, which causes a hearing problem. In the present embodiment, in consideration of the problem of the sound volume, the synthesis unit 123 adjusts the signal level of each of the initial reflected sound signal S121 and the reverberation sound signal S125 so that the sound volume is not unnatural. It is processing. The contents of this signal level adjustment processing will be described below.

A3: Level Adjustment Process First, the configuration of the synthesizing unit 123 will be described. FIG. 19 shows
It is a block diagram of the synthetic | combination part 123, The synthetic | combination part 123 is equipped with the level adjustment part 123A and the adder 123C. Of these, the level adjusting unit 123A includes two amplifiers 123A-
It is composed of 1,123A-2. Amplifier 123A
-1 includes an initial reflected sound signal S121 and an amplifier 123A-2.
Is supplied with a reverberation sound signal S125, and the entire signal is amplified at a predetermined amplification factor.

In this embodiment, under the control of the controller 2, the amplification factors G1 of the amplifiers 123A-1 and 123A-2 are based on the values of the impulse response waveform data.
G2 is predetermined. Below, the amplification factors G1 and G2
The method of determining the value of is explained. FIG. 20 shows the content of the data sequence D1 and the corrected data D1 ′ included in the predetermined period TD1 and the content of the data sequence D2 and the corrected data D2 ′ included in the predetermined period TD2 of the impulse response waveform data. Is schematically shown. For convenience of explanation, the time axis in FIG. 20 is displayed so as to correspond to the timing of the initial reflected sound signal S121 and the reverberation sound signal S125 supplied to the synthesis unit 123.

In FIG. 20, first, the level of the data of the data string D1 included in the minute period β (β <α) immediately before the end in the data string D1 or the sum value SUM of powers.
In 1 and the data sequence D2, the level of the data of the data sequence D2 included in the minute period β after the lapse of α time from the start of the data sequence D2, or the sum value SUM2 of powers can be obtained. Note that the data level or the total power value may be calculated based on the data value from which the DC component has been removed. The value S obtained in this way
From the values of UM1 and SUM2, amplifier 1
Amplification factors G1 and G of 23A-1 and amplifier 123A-2
2 is determined.

[0046] G2 / G1 = SUM1 / SUM2 ... (1)

FIG. 21 shows amplifiers 123A-1 and 123A when the amplification factors G1 and G2 determined from the above equation (1) are used.
3A-2 initial reflection sound signal S121 ′ and reverberation sound signal S1
25 'is a schematic view. In addition, the broken line indicates that the initial reflected sound signal S121 ′ and the reverberant sound signal S125 ′ are synthesized by the synthesis unit 1
23 shows the signal S123 synthesized by H.23. Compared to the signal S123 shown in FIG. 18 above, the signal level (volume) connection is more natural. As described above, even when the reverberation sound signal S125 is combined with the initial reflected sound signal S121 by the function of the level adjusting unit 123A, not only the sound quality but also the level (volume) of the acoustic signal is unnatural in terms of hearing. The effect that does not become a situation is played.

As described above, in the reverberation imparting apparatus 100 according to the present invention, among the sampling data of the impulse response waveform, the sampling data included in the predetermined period (for example, the period from the generation of the impulse sound to 0.2 seconds). Is convolved in the convolution operation unit 121 to generate an initial reflected sound signal S121 relating to the initial reflected sound. Therefore, with respect to the initial reflected sound that characterizes the acoustic space, the data content of the impulse response waveform can be faithfully reproduced, and sufficient acoustic characteristics can be expressed. Further, among the sampling data of the impulse response waveform, the convolution operation unit 122 convolves the sampling data included in another predetermined period (for example, 0.3 to 0.6 seconds), and the signal obtained as a result is attenuated in the filter 125. The reverberant sound signal S12 in which the rear reflected sound has a reverberant effect by repeatedly outputting while
5 can be generated. Therefore, it is not necessary to convolve all the sampling data of the impulse response waveform, so that enormous hardware resources are not required.
Here, in the synthesis unit 123, the initial reflected sound signal S121 related to the initial reflected sound and the reverberation sound signal S related to the rear reverberant sound.
Since a period Tα in which the two signals overlap each other is provided when 125 is combined, in the combined signal S123, the initial reflected sound signal S121 to the reverberation sound signal S12 are included.
There is no sudden change to 5 and the combined signal S1
When 23 is output as an acoustic signal, an unnatural situation in which the sound quality suddenly changes does not occur. Further, in the period Tα, the convolution operation units 121 and 122 correct the multiplication coefficient so that the initial reflected sound signal S121 and the reverberation sound signal S125 are cross-faded. Therefore, in the synthesized signal S123, the reverberation sound signal S125 is changed from the initial reflected sound signal S121.
Therefore, when the synthesized signal S123 is output as an acoustic signal, there is an effect that the change in sound quality becomes smoother. Further, in the synthesizing unit 123, the level of the two signals S121 and S122 is adjusted, and then the synthesizing process is performed. Therefore, the composite signal S
When 123 is output as an acoustic signal, not only the sound quality but also the change in the volume is audibly smooth and natural.

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 above embodiment, in the sampling data of the same impulse response waveform, the data strings included in the predetermined periods T1 and T2 are treated as the data strings D1 and D2.
Not limited to this, the data strings D1 and D2 may be based on sampling data of different impulse response waveforms. Even in such a case, as in the above-described embodiment, the signal S123 output from the synthesizing unit 123 has a smooth connection in both sound quality and volume, so that a reverberation characteristic that is natural in terms of hearing can be imparted.

(Modification 2) The method of determining the amplification factors G1 and G2 of the level adjusting unit 123A can be modified arbitrarily. For example, as shown in FIG. 22, the level of the data of the data sequence D2 included in the minute period γ (γ <α) immediately after the start of the data sequence D2, or the total power value SU.
In M4 and the data string D1, the level of the data of the data string D1 included in the minute period γ immediately before the lapse of T124 time from the start of the data string D1, or the total value S of the powers
It is also possible to obtain UM3 and determine the amplification factors G1 and G2 by the following equation.

[0052] G2 / G1 = SUM3 / SUM4 ... (2)

Also by this method, the connection of the data level values of the synthesized signal S123 obtained by synthesizing the initially reflected sound signal S121 relating to the initial reflected sound and the reverberation sound signal S125 relating to the rear reflected sound is natural. Therefore, the same effect as that of the above-described embodiment can be obtained.

(Modification 3) The configuration of the filter 125 can be modified arbitrarily. For example, the phase of the signal output from the filter 125 may be adjusted by adding a plurality of all-pass filters (APFs) in series or in parallel after the filter 125. Alternatively, a multi-tap delay may be added after the filter 125 to adjust the time density of the signal output from the filter 125.

(Modification 4) The configuration itself of the reverberation imparting apparatus 100 can be arbitrarily modified. For example, as shown in FIG. 24, the reverberation sound signal S125 relating to the rear reflection sound may be generated by repeatedly outputting the initial reflection sound signal S121 output from the convolution operation unit 121 while repeatedly attenuating the filter 125. . This modification has an advantage that the convolution operation unit 122 according to the above-described embodiment can be omitted as a component, while the filter 1
It is necessary to make the shape of the output reverberation sound signal S125 of No. 25 into which the fade-in process has been performed. Therefore, filter 1
The value of the filter coefficient in 25 is adjusted to an appropriate value, and the signal level of the reverberation sound signal S125 gradually increases from 0 immediately after the start of the output reverberation sound signal S125 of the filter 125 (the portion of α time). In other words, the signal shape may be such that the fade-in processing has been performed. Even in the case of such a configuration of the present modified example, in the present embodiment as well as the above-described first embodiment,
When the initial reflected sound signal S121 related to the initial reflected sound and the reverberant sound signal S125 related to the rear reverberant sound are combined, it is possible to perform a combining process in which a crossfade is performed for a predetermined period. Therefore, in the combined signal S123, the connection of signals (data) does not deteriorate, and a reverberation effect that is natural in terms of music can be added.

(Modification 5) Further, as shown in FIG. 25, the configuration of the reverberation imparting apparatus 100 is such that the rear reflection sound signal S122 output from the convolution operation unit 122 and the initial reflection output from the convolution operation unit 121. You may make it follow the sound signal S121. More specifically, the rear reflected sound signal S122 output from the convolution operation unit 122 is supplied to the adder 128 via the delay unit 127,
The initial reflected sound signal S output from the convolution operation unit 121
121 follows. In this case, the convolution operation unit 121
As the data sequence to be convolved in the convolution operation unit 122, data that is temporally continuous in the original impulse response is used. Then, the delay time T of the delay device 127
If 127 is made equal to the time length T121 of the initial reflected sound signal S121, the signal S128 in which the rear reflected sound signal S122 is directly followed by the rear reflected sound signal S122 is
It is output from the adder 128. Here, the initial reflected sound signal S121 and the rear reflected sound signal S122 are signals obtained by directly convolving the sampling data of the original impulse response waveform, and are effective signals that faithfully reproduce the data content of the impulse response waveform. It will be used. However, in the present modification, it is necessary to perform data correction only on the data sequence D2 to be convolved in the convolution operation processing unit 122. Specifically, the data string D2
The data correction may be performed so that the data value gradually decreases in the data included in the predetermined time immediately before the end, that is, the data value fades out.
On the other hand, similarly to the fourth modification, the filter coefficient value in the filter 125 is adjusted to an appropriate value, and the filter 1
Immediately after the start of the output reverberation sound signal S125 of No. 25, the signal level of the reverberation sound signal S125 may be gradually increased from 0, that is, the signal shape may be such that the fade-in process is performed. By performing such a process, similar to the above-described embodiment, in the synthesizing unit 123, when performing the synthesizing process of the signal S128 and the reverberation sound signal S125, the synthesizing process is performed so as to perform the crossfade for a predetermined period. You can Therefore, the composite signal S
In 123, the connection of signals (data) is not deteriorated, and a reverberation effect that is natural to the listener can be provided.

(Modification 6) In the above-described embodiment, the sampling data of the impulse response waveform is stored in advance in the reverberation data memory 107, but the sound field simulation program is stored in advance in the ROM (Read (not shown) in the controller 2). It may be stored in a memory (Only Memory) so that the user can program the impulse response waveform of an arbitrary acoustic space.

(Modification 7) A plurality of impulse response waveform data may be stored in the reverberation data memory 107. In this case, as shown in FIG. 23, 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 1 to select a desired acoustic space.

(Modification 8) In the above embodiment, the reverberation applying apparatus 100 equipped with the reverberation applying function has been described, but in addition to this, even a device such as a mixer or reverb equipped with the reverberation applying function is provided by the present invention. The application of is naturally possible.

(Modification 9) A recording medium for recording the program according to the present invention is also arbitrary, and for example, a semiconductor memory,
CD-ROM (Compact Disc-Read Only Memory), C
Optical disc such as D-R (Compact Disc-Recordable), M
Examples thereof include magneto-optical disks such as O (Magneto Optical Disk) and MD (Mini Disc), floppy (registered trademark) disks, magnetic disks such as hard disks. Also,
The method for installing such a program is also arbitrary, and may be installed in the reverberation imparting apparatus 100 using the recording medium described above. A method using so-called net distribution, which is installed in 100, may be used.

[0061]

As described above, according to the present invention,
It is possible to express a sufficient sound field effect with a simple configuration.

[Brief description of drawings]

FIG. 1 is a reverberation imparting apparatus 100 according to an embodiment of the present invention.
It is a block diagram of.

FIG. 2 is a configuration diagram of a DSP 5 of the reverberation imparting apparatus 100.

FIG. 3 is a diagram showing sampling data of impulse response waveforms.

FIG. 4 is a reverberation imparting apparatus 100 according to an embodiment of the present invention.
6 is a configuration diagram of a reverberation imparting unit 120 of FIG.

FIG. 5 is a convolution operation unit 12 of the reverberation imparting apparatus 100.
It is a block diagram of 1.

FIG. 6 is a convolution operation unit 12 of the reverberation imparting apparatus 100.
It is a block diagram of 2.

FIG. 7 is a filter 125F- of the reverberation applying apparatus 100.
It is a figure which shows the shape of the output signal of No. 1 typically.

FIG. 8 is a diagram for explaining the signal processing content of a filter 125 of the reverberation applying apparatus 100.

FIG. 9 is a diagram showing a signal timing supplied to the synthesis unit 123 of the reverberation imparting apparatus 100.

FIG. 10 is a data sequence D in the reverberation applying apparatus 100.
FIG. 6 is a diagram for explaining the content of correction for 1.

FIG. 11 is a data sequence D in the reverberation applying apparatus 100.
It is a figure for demonstrating the content of correction with respect to 2.

FIG. 12 is a diagram schematically showing data contents stored in a reverberation data memory 107 of the reverberation imparting apparatus 100.

FIG. 13 is a convolution operation unit 1 of the reverberation applying apparatus 100.
It is a figure which shows the output initial reflected sound signal S121 of 21 typically.

FIG. 14 is a convolution operation unit 1 of the reverberation imparting apparatus 100.
22 is a diagram schematically showing an output rear reflected sound signal S122 of FIG.

15 is a diagram schematically showing an output reverberation sound signal S125 of a filter 125 of the reverberation applying apparatus 100. FIG.

FIG. 16 is a diagram for explaining the operation of the synthesis unit 123 of the reverberation imparting apparatus 100.

FIG. 17 is a diagram for explaining the operation of the synthesizing unit 123 of the reverberation imparting apparatus 100.

FIG. 18 is a diagram for explaining the operation of the synthesis unit 123 of the reverberation imparting apparatus 100.

FIG. 19 is a configuration diagram of a synthesizer 123 of the reverberation imparting apparatus 100.

FIG. 20 is a level adjusting unit 12 of the reverberation applying apparatus 100.
It is a figure for demonstrating the determination method of the amplification factor of 3A.

FIG. 21 is a diagram showing a combined signal S123 of the combining unit 123 of the reverberation imparting apparatus 100.

FIG. 22 is a diagram for explaining a modified example of the present invention.

FIG. 23 is a diagram for explaining a modified example of the present invention.

FIG. 24 is a diagram for explaining a modified example of the present invention.

FIG. 25 is a diagram for explaining a modified example of the present invention.

[Explanation of symbols]

100 ... Reverberation imparting device, 1 ... Operation unit, 2 ...
Controller, 3 ... Input section, 4 ... Output section, 5 ... D
SP, 6 ... Sound source, 9 ... Bus, MIC ... Microphone, SP ... Speaker, 107 ... Reverberation data memory,
120 ... Reverberation adding section, 121, 122 ... Convolution calculating section, 123 ... Combining section, 123A ... Level adjusting section,
123C ... Adder, 124 ... Delay device, 125 ... Filter.

Continued front page    (72) Inventor Kenichi Tamiya             Yamaha stock, 10-1 Nakazawa-machi, Hamamatsu-shi, Shizuoka             In the company (72) Inventor Hiroaki Fujita             Yamaha stock, 10-1 Nakazawa-machi, Hamamatsu-shi, Shizuoka             In the company F-term (reference) 5D108 AA17 AA18 AB08 AB09 AB19                       AC06 AC07 AD05

Claims (12)

[Claims] 1. A first signal processing means for generating a first reflected sound signal by convolving a first data sequence corresponding to a first predetermined time of an impulse response with an acoustic signal to be processed. , The first signal after performing a predetermined signal processing on the acoustic signal
Second signal processing means for generating a second reflected sound signal subsequent to the reflected sound signal, the first reflected sound signal generated by the first signal processing means, and the second signal processing means. Means for adding and outputting the second reflected sound signal generated by the output means for outputting the second reflected sound signal with a delay of a time shorter than the first predetermined time. A reverberation imparting device comprising: 2. The reverberation applying apparatus according to claim 1, wherein the second signal processing means uses the recursive filter for the first reflected sound signal generated by the first signal processing means. A reverberation imparting device, which is means for generating the second reflected sound signal by repeatedly outputting. 3. A first signal processing means for generating a first reflected sound signal by convolving a first data string corresponding to a first predetermined time of an impulse response with an acoustic signal to be processed. , The first data sequence corresponding to the second predetermined time of the impulse response is convoluted with the acoustic signal, and the convolutionally generated signal is repeatedly output using a recursive filter. Second signal processing means for generating a second reflected sound signal subsequent to the reflected sound signal, the first reflected sound signal generated by the first signal processing means, and the second signal processing means. Means for adding and outputting the second reflected sound signal generated by the output means for outputting the second reflected sound signal with a delay of a time shorter than the first predetermined time. Equipped with A reverberation applying device characterized by: 4. A signal obtained by convolving a first data sequence corresponding to a first predetermined time of an impulse response with respect to an acoustic signal to be processed corresponds to a second predetermined time of the impulse response. First signal processing means for generating a first reflected sound signal by concatenating a signal obtained by convolving the second data sequence, and the second data in the first signal processing means. Second signal processing means for generating a second reflected sound signal following the first reflected sound signal by repeatedly outputting a signal obtained by convoluting the columns using a recursive filter; Means for adding and outputting the first reflected sound signal generated by the first signal processing means and the second reflected sound signal generated by the second signal processing means, An output means for delaying and outputting the second reflected sound signal by a time shorter than a value obtained by adding the first predetermined time and the second predetermined time. . 5. The reverberation imparting apparatus according to claim 1, wherein the output unit cross-fades the first reflected sound signal and the second reflected sound signal and outputs the cross reflected signal. A reverberation applying device characterized by being performed. 6. The reverberation imparting apparatus according to claim 1, wherein the output means outputs one or both of the first reflected sound signal and the second reflected sound signal. A reverberation applying device having a signal level adjusting means for adjusting a level. 7. The reverberation imparting apparatus according to claim 6, wherein the signal level adjusting means includes a level value of a signal included in the first predetermined time of the first reflected sound signal, and the second level value. A reverberation applying apparatus, which adjusts a signal level according to a level value of a signal included in a predetermined time of a reflected sound signal. 8. A first signal processing step of generating a first reflected sound signal by convolving a first data sequence corresponding to a first predetermined time of an impulse response with an acoustic signal to be processed, , The first signal after performing a predetermined signal processing on the acoustic signal
Second signal processing step of generating a second reflected sound signal following the reflected sound signal of No. 1, the first reflected sound signal generated by the first signal processing step, and the second signal processing step A step of adding and outputting the second reflected sound signal generated by the step of outputting the second reflected sound signal after delaying the second reflected sound signal by a time shorter than the predetermined time. A reverberation imparting method comprising: 9. A first signal processing step of generating a first reflected sound signal by convolving a first data string corresponding to a first predetermined time of an impulse response with an acoustic signal to be processed, , The first data sequence corresponding to the second predetermined time of the impulse response is convoluted with the acoustic signal, and the convolutionally generated signal is repeatedly output using a recursive filter. Second signal processing step of generating a second reflected sound signal following the reflected sound signal of No. 1, the first reflected sound signal generated by the first signal processing step, and the second signal processing step A step of adding and outputting the second reflected sound signal generated by the step of outputting the second reflected sound signal after delaying the second reflected sound signal by a time shorter than the first predetermined time. Equipped with A reverberation applying method characterized by: 10. A first impulse response to an acoustic signal to be processed by a computer.
A first signal processing means for generating a first reflected sound signal by convolving a first data string corresponding to a predetermined time of, and the first signal processing means for applying a predetermined signal processing to the acoustic signal.
Second signal processing means for generating a second reflected sound signal subsequent to the reflected sound signal, the first reflected sound signal generated by the first signal processing means, and the second signal processing means. Means for adding and outputting the second reflected sound signal generated by the output means, which outputs the second reflected sound signal with a delay of a time shorter than the first predetermined time. Program to function as. 11. A first impulse response to a sound signal to be processed by a computer.
A first signal processing means for generating a first reflected sound signal by convolving a first data string corresponding to a predetermined time of, and a second predetermined time of the impulse response to the acoustic signal. A second signal for generating a second reflected sound signal following the first reflected sound signal by convolving the second data string and repeatedly outputting the convolved generated signal using a cyclic filter. Processing means, means for adding and outputting the first reflected sound signal generated by the first signal processing means and the second reflected sound signal generated by the second signal processing means A program for causing the second reflected sound signal to function as an output unit that delays and outputs the second reflected sound signal by a time shorter than the first predetermined time. 12. A computer-readable recording medium in which the program according to claim 10 or 11 is recorded.