JP2000097763A - Method for collecting impulse response, sound effect adder, and record medium - Google Patents

Abstract

PROBLEM TO BE SOLVED: To readily obtain high-quality reverberated sounds in response to a surround system. SOLUTION: A sound source 102 is installed at a predetermined position of a stage part 101A. Impulse response measuring microphones 103FL, 103FR, 103RL, 103RR are each installed at predetermined positions in response to a surrounding system in a seat part 101B. Measuring TSP signals reproduced by the sound source 102 are each collected by four mikes, and each of four channels is converted into digital signals by an A/D converter 93', and is supplied to an impulse response collector 97'. In the device 97', measured data of the four channel are each divided by inverse characteristics of the TSP signals, thereby obtaining impulse response data at each of the four channels. The impulse response data at each of the four channels are recorded in, e.g. CD-ROMs. Audio signals are each folded in by the impulse response data of the four channels, thereby obtaining reverberated sounds in response to the surround system.

Description

DETAILED DESCRIPTION OF THE INVENTION

[0001]

The present invention relates to an impulse response collecting method and a sound effect adding apparatus which are particularly suitable for a surround effect when adding a reverberant sound based on an impulse response of an actual device or space. It relates to a recording medium.

[0002]

2. Description of the Related Art One of devices for adding a sound effect to an audio signal is a reverberation device (reverberator). This reverberation adding apparatus is often used in a recording studio, for example, to add reverberation to an audio signal to give a sound a spaciousness or depth. By adding reverberation to a sound recorded in a studio or the like, it is possible to give an effect as if it were actually played in a hall or a more special effect.

[0003] In the past, reverberation was added by recording in a place such as a hall where reverberation could be obtained.
Alternatively, it has been performed using a device such as an iron plate echo that uses a vibration of an iron plate to obtain a reverberant effect. These effects are realized electrically in recent reverberation adding apparatuses. Further, in recent years, with the development of digital signal processing technology, devices for digitally synthesizing reverberation have become widespread.

When a reverberation sound is added by digital processing, for example, a cyclic digital filter is used. The input digital audio signal is circulated while being attenuated, and reverberation is generated. This is mixed with the original digital audio signal. In practice, an initial reflected sound is added to a position delayed by a predetermined period from the original sound, and a reverberation sound is added after a predetermined period. The delay time of the reverberation sound with respect to the original sound is called pre-delay.
Reverberation time and sub-reverberation can be added and fine level adjustments can be made, making it possible to create a wide range of sounds.

[0005] By the way, a reproduction method called a surround method, in which a sound field from the rear is added to a sound field reproduction from the left and right sound sources from the front, is becoming widespread. The surround system is used especially for reproducing the sound field of a movie theater. For example, left and right channels (FL, FR) arranged in front of the listener and left and right channels arranged in the rear of the listener. Channels (RL, RR) are provided, and one channel (C) is arranged at the center in front of the listener. The C channel is for reproducing, for example, dialogue in movie sound. further,
One channel for reproducing an extremely low frequency band is arranged at an arbitrary position. In the configuration of 5 channels + ultra-low frequency channel, the transmission amount of information in the ultra-low frequency band is only 1/10 of that of the other channel, so that the channel is expressed as 5.1 channels as a whole.

[0006] Not only movies, but also regular music performances,
By reproducing the sound recorded in the surround system in a corresponding system, it is possible to enjoy a more realistic performance. In the case of music performance, FL, FR, R
-L and R-R channels are used, and C channel is not used.

[0007]

When recording in the surround system, it is necessary to add a reverberant sound corresponding to a sound field formed by four channels. Conventionally, two monaural input / stereo output reverberation sound adding devices are often used. In addition, a digital reverberation device is used to obtain higher quality reverberation.

For example, when recording one sound source supposed to be on the stage, one reverberant sound adding device records FL, FL, supposed on the stage side with respect to the listener.
The reverberation corresponding to the F-R channel is added, and the reverberation sound adding device adds the reverberation corresponding to the RL and RR channels assumed backward to the listener.

Since the reverberation differs depending on the position, these two
Each of the reverberation sound adding devices has a different setting. As described above, in the related art, when performing recording in accordance with the surround system, it is necessary to use two reverberation sound adding devices, so that there is a problem that the operation is inconvenient.

Further, the reverberation sound in an actual hall or the like has a more complicated waveform due to various reflections and interferences depending on the shape of the hall and the position of the sound source. However, as described above, in the method of filtering the original digital audio signal, there is a problem that an artificial impression is inevitably avoided since only an attenuated waveform is obtained.

On the other hand, for example, a sound source having a predetermined sound field
Must be recorded in stereo. In this case, a reverberation sound adding device corresponding to a stereo input is used. However, the conventional reverberation sound adding device that supports stereo input has a problem that the sound image is unnatural because a stereo sound image is artificially created.

To obtain more natural reverberation, recording in an actual hall may be considered. However, there was a problem that recording in the hall was extremely troublesome, such as loading equipment. Also, there was a problem that the hall could not be borrowed when it was desired. Furthermore, recording in the hall has a problem that know-how is required for setting a microphone and the like. Furthermore, there is a problem that it is necessary to stop an air conditioning facility or the like in order to reduce the background noise.

Accordingly, an object of the present invention is to provide an impulse response collecting method, a sound effect adding device, and a recording medium that can easily obtain a high-quality reverberation sound corresponding to a surround system.

[0014]

SUMMARY OF THE INVENTION In order to solve the above-mentioned problems, the present invention provides a method for collecting an impulse response used for generating a sound effect by convolving an impulse response. Measuring each of four or more spatially arranged measuring means of the spatial sound generated based on the above, and converting each of the measured sound into an impulse response. This is a method for collecting impulse responses.

The present invention also relates to a method of collecting impulse responses used for generating a sound effect by performing convolution of an impulse response, wherein the impulse response is generated based on a measurement signal at each of two points corresponding to a stereo system. Measuring the sound of the measured space with each of two or more spatially arranged measuring means, and converting each of the measured sound into an impulse response. This is a method for collecting impulse responses.

The present invention also provides a measuring step of measuring the sound in the space generated based on the measuring signal by each of four or more spatially arranged measuring means; And a conversion step of converting the data into an impulse response. The computer-readable recording medium has recorded thereon impulse response data.

Also, the present invention provides a measurement method in which two or more spatially arranged measuring means measure sound in a space generated based on a measuring signal at each of two places corresponding to a stereo system. And a step of converting each of the measured sounds into an impulse response. The recording medium is readable by a computer.

The present invention also provides a sound effect adding apparatus for adding a sound effect by convolving impulse response data to an input digital audio signal, wherein a plurality of impulse response signals respectively corresponding to a plurality of locations in the same space are provided. Reproducing means for reproducing a plurality of impulse response data from a recording medium on which data is recorded; input means for inputting a digital audio signal; and digital data input by the input means based on the plurality of impulse response data reproduced by the reproducing means. A sound effect adding apparatus comprising a plurality of convolution means for convolving an audio signal.

As described above, according to the present invention, the sound of the space generated based on the measurement signal is measured by each of four or more spatially arranged measuring means, and the measurement is performed. Since each of the obtained sounds is converted into an impulse response, an impulse response corresponding to the surround method can be collected.

Further, as described above, according to the present invention, the sound of the space generated based on the measurement signal at each of the two points corresponding to the stereo system is spatially arranged. Measuring with each of the more than one measuring means,
Since each of the measured sounds is converted into an impulse response, an impulse response corresponding to a stereo input and a stereo output can be collected.

As described above, according to the present invention, the sound of the space generated based on the measurement signal is measured by each of four or more spatially arranged measuring means. Since the impulse response data obtained by converting each of the measured sounds into an impulse response is recorded on the recording medium, it is possible to store and reproduce the impulse response corresponding to the surround in real space.

Further, as described above, according to the present invention, the sound of the space generated based on the measurement signal at each of the two locations corresponding to the stereo system is spatially arranged. Measuring with each of the more than one measuring means,
Since impulse response data obtained by converting each of the measured sounds into impulse responses is recorded on a recording medium, storing and reproducing impulse responses corresponding to stereo input and stereo output in a real space. Can be.

Further, as described above, the present invention according to claim 8 reproduces a plurality of impulse response data from a recording medium on which a plurality of impulse response data respectively corresponding to a plurality of locations in the same space are recorded, Since the input digital audio signal is convolved with a plurality of reproduced impulse response data, the impulse response in a real space can be reproduced.

[0024]

DETAILED DESCRIPTION OF THE PREFERRED EMBODIMENTS An embodiment of the present invention will be described below. The sound effect adding device in this embodiment is a reverberation adding device for adding a reverberant sound to an original sound composed of an input digital audio signal, and is an impulse obtained by collecting reverberation of an actual hall or the like. The input digital audio signal is convolved with the response data to obtain a reverberation sound to be added.

FIG. 1 shows a reverberation sound according to this embodiment in comparison with a reverberation sound obtained by a conventional recursive filter. FIG.
A reverberation sound according to the prior art shown in FIG. 1A is delayed from the original sound by a predetermined time to generate an initial reflection sound, and further delayed by a predetermined time to add a reverberation sound generated by a filter. The reverberation to be added is attenuated by a simple decay curve. In contrast, in this embodiment, the reverberation sound is generated by an impulse response based on actually recorded data, and therefore, as shown in FIG.
A reverberant sound that is not a simple attenuation curve and reflects the acoustic characteristics of an actual hall or the like can be obtained. Thereby, a more natural and high-quality reverberation sound can be obtained.

The present invention provides an impulse response collecting method for obtaining such a natural reverberation sound. In particular, the present invention provides a method of collecting impulse responses suitable for use in a surround system. FIG.
Shows an example of the configuration of the impulse response collection device 97 according to the present invention. In this example, the impulse response of the iron plate echo device 92 is measured. Impulse response collection device 97
Can be constituted by, for example, a personal computer.
This device 97 generates an impulse response measurement signal, outputs the signal to a measurement target, collects measurement results, and converts the measurement results into impulse response data. The impulse response data is stored, for example, as a file.

The measurement signal generator 90 generates a TSP (time stretch pulse) signal for measuring an impulse response. The TSP signal is a kind of sweep signal, and an impulse signal can be obtained by dividing the signal by a signal having an inverse characteristic. In order to measure the impulse response, it is more preferable to directly generate an impulse signal. However, since measurement is difficult, such a method is used.
The TSP signal generated by the measurement signal generator 90 is D /
The signal is converted into an analog signal via the A converter 91 and input to the iron plate echo device 92.

In the iron plate echo device 92, the input TS
A reverberation sound is generated by the P signal. This reverberation is L
These are output as analog audio signals of the (left) and R (right) channels. These outputs are converted into digital audio signals for the L and R channels by the A / D converter 93. The A / D converter 93 performs sampling at, for example, a sampling frequency of 48 kHz or 96 kHz and a quantization bit number of 24 bits.
As for the output of the A / D converter 93, each of the L and R channels is input to the impulse response collection device 97. The input signal is stored in, for example, a hard disk drive or a memory (not shown).

The reverberation time is defined as the time from when the sound stops until the sound pressure level attenuates by 60 dB. In this example, 6 dB is allocated to 1 bit in the quantization bit number of 24 bits.

The generation of the TSP signal by the measurement signal generator 90 is performed N times. The output signals for N times are synchronously added by the synchronous addition section 94. The synchronous addition is performed so that the output signals are aligned based on, for example, the timing of generation of the TS signal. By synchronously adding the N signals, only the reproducible signal is added and the noise component generated at random is not added, so that the S / N ratio can be improved. The S / N ratio is improved by (10 log N) dB. For example, the S / N ratio is improved by 12 dB when N = 16.

Each of the L and R channels of the synchronously added signal is supplied to an impulse response converter 95. The impulse response converter 95 divides the supplied signal by a signal having the inverse characteristic of the TSP signal. As a result, the TSP signal is converted into an impulse signal, and the measurement result is converted into an impulse response based on the reverberation generated by the impulse signal. The impulse response data is
It is a peak value obtained at intervals corresponding to the sampling frequency. The signal sampled by the A / D converter 93 with a 24-bit quantization bit number has a 32-bit quantization bit number after conversion.

L output from the impulse response conversion unit
The channel impulse response data 96L and the R channel impulse response data 96R are stored in a CD-ROM.
And MO are recorded on a predetermined recording medium. The impulse response collection device 97 may be provided with an interface such as Ethernet, and supplied to the outside via a network.

FIG. 3 shows an example in which an impulse response is collected in a hall. The hall 101 has a stage 101
A and a seat 101B. Stage unit 101A
The sound source 102 is set at a predetermined position. Sound source 102
Are, for example, dodecahedral speakers provided with speakers in 12 different directions on a spherical surface. Microphones 103L and 103R respectively corresponding to the L and R channels are installed at predetermined positions in the customer seat 101B.

The TSP signal output from the impulse response collection device 97 is converted into an analog signal by the D / A converter 91, amplified by the amplifier 100, and reproduced by the sound source 102 as sound. This reproduced sound is transmitted to the microphone 103L.
And 103R. The outputs of the microphones 103L and 103R are sampled by the A / D converter 93 at a predetermined sampling frequency and a predetermined number of quantization bits, respectively, and are converted into L and R channel digital audio signals, which are supplied to the impulse response collection device 97. The processing in the impulse response collection device 97 is exactly the same as the processing in the iron plate echo device 92 described above.

In this case, the position of the sound source 102 is variously changed, and the impulse response is collected. Also, the sound source 10
The loudspeaker used as 2 is collected by changing its brand and the like in various ways. Similarly, the microphone 103
L and 103R are also recorded in various positions and brands. Thus, in one hole 101, a plurality of data are collected. These may be selectable as variations of the reverberation sound, for example, when adding the reverberation sound.

On the other hand, the impulse response data 96L and 96R obtained by the impulse response converter 95 can be processed. FIG. 4 schematically shows the flow of processing when processing impulse response data. The impulse response data 110 is subjected to a processing 111. FIG. 5 shows an example of the processing 111. As shown in an example in FIG. 5A, the data has a system delay due to sound propagation (“A” part in the figure). In the processing 111, the value of the system delay unit A is

It is fixed to [0], and the noise in this part is removed.

In the latter half of the data, the end of the data is

A fade-out process is performed to converge to [0]. This fade-out processing also removes noise in the second half minute level signal portion. 5B and 5C show an example of the fade-out processing.

FIG. 5B shows an example in which fade-out processing is performed based on an exponential function of attenuation. For example, assuming that the original impulse response is h (n), the fade-out function is F
₀ (n). n represents a point of the impulse response data. Note that the points of the impulse response data
The points of the sampling points of the digital audio signal correspond to each other. At this time, the F ₀ (n), if n ≦ 0, a F ₀ (n) = 1. On the other hand, n
If> 0, F ₀ (n) is an exponential function of the attenuation as shown in FIG. 5B.

The output data x (n) is given by the following equation (1): x (n) = h (n) · F ₀ (na) (1) The value a represents the position of the direct sound in the original impulse response in terms of the number of samples. in this way,
Fade-out occurs after the position of the direct sound. This is because if the fade-out starts at the same position as the direct sound, that is, when n = 0, the level of the direct sound itself also decreases.

Note that the fade-out function is not limited to the exponential function of attenuation. For example, as shown in FIG.
A linear attenuation characteristic may be used.

Further, the number of points of the impulse response data can be adjusted by the fade-out so as to match the processing capability of a reverberation device that actually adds a reverberant sound to an audio signal using the data. That is, the number of points of the impulse response data is limited to a predetermined value, for example, 256 k points (262, 144 points: 256 k points with fractions omitted. The expression of the value of 2 ⁿ is the same hereinafter). Sometimes
For example, as shown in FIG. 4A, the fade-out starts at the point of 128 k points, and the data starts to fade at the point of 256 k points.

[0].

As the processing 111, in addition to the above, level adjustment and the like are also performed. The processed impulse response data is used for the FI at the time of convolution by the FIR filter.
As the R filter coefficient 112, for example, a CD-ROM 45
Will be recorded.

FIG. 6 schematically shows an example of the configuration of a reverberation adding apparatus that performs convolution using the impulse response data created in this way. A digital audio signal to which a reverberation sound is to be added is input from an input terminal 120. The input data is supplied to a multiplier 126 and
It is delayed by the predelay 121 and is given a predelay. The output of the predelay 121 is supplied to the convolution processing unit 122.

The convolution processing section 122 includes an FIR filter (filter 122L) for each of the L and R channels.
And a filter 122R). The impulse response data 96L and 97R created by the impulse response collecting device 97 described above are supplied from the terminals 123L and 123R as FIR filter coefficients of the corresponding channels. These impulse response data 96L and 96L
R is obtained, for example, by reading from a CD-ROM (not shown).

The filters 122L and 122R convolve the input digital audio signal with the impulse response data 96L and 97R. As a result of this convolution, the impulse response data 96
A reverberation based on L and 96R is generated. The outputs of the filters 122L and 122R are
4L and 124R.

The multipliers 124L and 124R, the multiplier 126 described above, and the adders 128L and 128R constitute a mixer for the original sound (dry component) and the reverberant sound (wet component). Depending on the ratio of the original sound and the reverberant sound supplied to the terminals 127 and 125, respectively, the multiplier 126
The input digital audio signal and the output of the convolution processor 122 are adjusted by the multipliers 124L and 124R, and these signals are added by the adders 128L and 128R, and the output of the L channel is output to the output terminal 129L.
The output of the R channel is led out to the output terminal 129R.

Further, the present invention provides a method of collecting impulse responses corresponding to the surround system. FIG. 7 shows an example in which an impulse response is collected so as to correspond to the surround system. The collection of the impulse response is performed in the same manner as described above.
01B. A sound source 102 composed of, for example, a dodecahedron speaker is installed at a predetermined position of the stage 101A.

The seat 101B has four microphones 1 corresponding to the surround system.
03FL, 103FR, 103RL and 103RR are set at predetermined positions. That is, the microphone 10
3FL and 103FR correspond to the L and R channels (FL and FR channels) on the front side of the customer seat 101B, respectively. The microphone 10
3RL and 103RR respectively correspond to L and R channels (RL and RR channels) on the rear side of the passenger seat 101B.

The microphones 103FL, 103
As an example, the positions of FR, 103RL, and 103RR are such positions that a sufficient surround effect can be obtained for the listener, assuming that the listener is seated at the optimum position of the stage 101A with respect to the stage 101A. It is said.

The microphones 103FL, 103FR,
The output of each of 103RL and 103RR is A /
It is supplied to a D converter 93 '. The A / D converter 93 ′
The above-described A / D converter 93 is expanded so that signal processing for four channels can be performed. A / D converter 9
At 3 ', sampling is performed at a predetermined sampling frequency and a predetermined number of quantization bits, FL (front left channel), FR (front right channel), RL
(Rear left channel) and RR (Rear right channel) digital audio signals are supplied to the impulse response collection device 97 '.

The impulse response collecting device 97 'is obtained by expanding the above impulse response collecting device 97 so that signal processing for four channels can be performed. In the impulse response collecting apparatus '97, processing similar to the processing in the iron plate echo apparatus 92 described above is performed by FL, FR, RL, and R-
The impulse response data 96F-L, 96F-R, 96R for the four channels R
-L and 96R-R are obtained (not shown).

In this case as well, the sound source 102
The impulse response is collected at various positions. In addition, the speaker used as the sound source 102 is collected by changing its brand and the like in various ways. Similarly, 4
Book microphones 103FL, 103FR, 103R
L and 103RR are also recorded in variously changed positions and brands. Thus, in one hole 101, a plurality of data are collected. These may be selectable as variations of the reverberation sound, for example, when adding the reverberation sound.

Also in this case, the impulse response data can be processed in the same manner as described above. The processed impulse response data is converted to a CD-R
Recorded in OM45.

FIG. 8 shows an example of the configuration of a reverberation adding apparatus for adding reverberation sound corresponding to a surround method to input data based on impulse responses collected based on the method shown in FIG. A digital audio signal to which a reverberation sound is to be added is input from an input terminal 130. The input data is supplied to a multiplier 136F to obtain a dry component for each of the FL, FR, RL, and RR channels.
L, 136FR, 136RL, and 136RR, and predelays 131FL, 131FR, 131RL, and 131RL to obtain wet components of the FL, FR, RL, and RR channels.
RR.

In this example, each of the FL, FR, RL and RR channels is independently subjected to the same processing. Hereinafter, the processing for these four channels is referred to as F
-L channel will be described as a representative. The input data of the FL channel is delayed by the predelay 131FL and is given a predelay. Predelay 313F
The output of L is supplied to an FIR filter 132FL that performs a convolution process of the impulse response data.

In the FIR filter 132FL, as the filter coefficient, the impulse response data 9 of the FL channel prepared by the impulse response
6FL is supplied from the terminal 133FL. These impulse response data 96F-L are stored in, for example, a CD-ROM 4
5 (not shown).

In the FIR filter 132F-L, the input digital audio signal is convolved by the impulse response data 96F-L. As a result of this convolution, a reverberation sound based on the impulse response data 96F-L is generated. The output of the FIR filter 132F-L is supplied to a multiplier 134.

Multiplier 134FL and multiplier 13 described above
The 6FL and the adder 138FL constitute a mixer of the original sound (dry component) and the reverberant sound (wet component) of the FL channel. Depending on the ratio of the original sound and the reverberant sound supplied to the terminals 137 and 135, respectively, the multiplier 13
The input digital audio signal directly input from the input terminal 130 and the output of the FIR filter FL 132 are adjusted by the 6 FL and the multiplier 134 FL, and the adder 1
At 38FL, these signals are added. The addition output is
The output of the FL channel is output to the output terminal 139FL.

The same processing is performed on the other three channels as on the FL channel. For example, the input data of the F-R channel is subjected to convolution processing by the impulse response data 96FR of the corresponding channel by the filter 132FR, and the ratio of the dry component to the wet component is determined by the mixer including the multipliers 137FR and 136FR and the adder 138FR. Is adjusted and output to the output terminal 139FR as the output of the FR channel.

Thus, FL, FR, RL
A reverberation sound based on the impulse response data of the corresponding channel is added to each of the R and R channels. The listener can reproduce the sound of the hall 101 in a surround system. For example, by listening at a predetermined position during reproduction, the listener can enjoy the effect as if he were in the customer seat 101B of the hall 101.

FIG. 9 shows an example in which an impulse response is collected so as to correspond to a stereo input. In this case, sound sources 102L and 102R, which are, for example, dodecahedral speakers, corresponding to the L and R channels, respectively, are mounted on the stage 101A. Sound source 102
L is at a predetermined position on the left side from the customer seat 101B,
The sound sources 102R are respectively set at predetermined positions on the right side as viewed from the front. On the other hand, two microphones 103F-L and 103F corresponding to the L and R channels, respectively, are provided in the customer seat portion 101B, as in the example shown in FIG.
The FR is set at a predetermined position.

[0062] The TSP signal from the amplifier 100 is
It is supplied to the sound source selected by the selection unit 104 among the 02L and 102R. The selection unit 104 may be a switch, or may directly switch the connection by attaching and detaching a cable.

The impulse response is collected for each of the L and R channels. By the selection of the selection unit 104, for example, the sound source 102L is selected. Sound source 102
L reproduces the TSP signal, and the microphone 103F-
The reproduced sound is recorded in L and 103F-R. The output of the microphones 103F-L and 103F-R is A
The signal is supplied to the impulse response collection device 97 via the / D converter 93, is subjected to impulse response conversion processing, and the L and R channel impulse response data 96L / FL and 96L / FR when the sound source is the L channel. Is obtained.

The impulse response is similarly collected for the sound source 102R, and the TSP signal reproduced by the sound source 102R by the selection of the selection unit 104 is recorded by the microphones 103F-L and 103F-R. Then, by the processing in the impulse response collection device 97, the impulse response data 96R / FL and 96R / FR of the L and R channels when the sound source is the R channel are obtained.

The impulse response data 96L / FL, 96L / FR, 96R / FL obtained in this manner.
And 96R / FR are recorded on the CD-ROM 45 after being subjected to, for example, predetermined processing.

FIG. 10 shows an example of the configuration of a reverberation adding apparatus for adding reverberation sound in stereo to data input in stereo based on impulse responses collected based on the method shown in FIG. Input data of the L channel is supplied from the input terminal 140L. Also, input data of the R channel is supplied from the input terminal 140R.

First, the processing of the L channel will be described. The input data supplied from the input terminal 140L is supplied to a multiplier 146L that adjusts the ratio of the dry component, and is also supplied to predelays 141LL and 141LR, and is independently supplied with a predelay. The outputs of the predelays 141LL and 141LR are respectively applied to a filter 142 for convolving the impulse response.
LL and 142LR.

The filter 142LL performs a convolution process on the data supplied from the predelay 141LL based on the impulse response data 96L / FL supplied from the terminal 143LL. The output of the filter 142LL is a multiplier 144LL for adjusting the ratio of the wet component.
Supplied to

Similarly, the output of the predelay 141LR is subjected to convolution processing by the filter 142LR using the impulse response data L / FR supplied from the terminal 143LR. The output of the filter 142LR is supplied to a multiplier 144LR that adjusts the ratio of the wet component.

Next, the processing of the R channel will be described. The input data supplied from the input terminal 140R is supplied to a multiplier 146R that adjusts the ratio of the dry component, and is also supplied to predelays 141RL and 141RR, each of which is independently supplied with a predelay. The outputs of the predelays 141RL and 141RR are respectively applied to a filter 142 for convolving the impulse response.
RL and 142RR.

The filter 142RL performs a convolution process on the data supplied from the predelay 141RL based on the impulse response data 96R / FL supplied from the terminal 143RL. The output of the filter 142RL is a multiplier 144RL for adjusting the ratio of the wet component.
Supplied to

Similarly, the output of the predelay 141RR is subjected to convolution processing by the filter 142RR using the impulse response data R / FR supplied from the terminal 143RR. The output of the filter 142RR is supplied to a multiplier 144RR that adjusts the ratio of the wet component.

Multipliers 144LL, 144LR and 14
6L adjusts the ratio of the dry component and the wet component to the input of the L channel. The ratio of the dry component of the L channel is given from the terminal 147L to the multiplier 146L. Also, terminals 145LL and 145LR
Therefore, the ratio of the wet component is independently given to each of the multipliers 144LL and 144LR. The outputs of multipliers 146L and 144LL are added to adder 148L
Supplied to The output of a multiplier 144RL, which will be described later, is further supplied to the adder 148L. Adder 148L
, These three outputs are added, and the added output is output to the output terminal 149L as the output of the L channel.

On the other hand, multipliers 144RR, 144RL and 146R adjust the ratio of the dry component and the wet component to the input of the R channel. The ratio of the dry component of the R channel is given from the terminal 147R to the multiplier 146R. Also, terminals 145RR and 14
From 5RL, the ratio of the wet component is independently given to each of multipliers 144RR and 144RL. The output of the multipliers 146R and 144RR is the adder 1
48R. The output of the above-described multiplier 144LR is further supplied to the adder 148R. Adder 1
48R, these three outputs are added, and the added output is output to the output terminal 149R as the output of the R channel.

As described above, in the case of the stereo input, the L and R channel input data correspond to the L and R channels generated by the sound sources 102L and 102R, respectively.
And R channel reverberations are added. Then, the output is mixed for each channel of the reverberation sound. Since a reverberation sound is added to each of the input L and R channels by adding the impulse responses actually collected in stereo for each of the L and R channels, a natural stereo sound image is obtained.

As shown by the dotted lines in FIG. 9, two microphones 103R-L and 1
03R-R may be installed. By doing so, it is possible to perform measurement corresponding to the surround method in addition to the stereo method. In this method, four microphones 10 are provided for each of the L and R channel sound sources.
3F-L, 103F-R, 103R-L and 103R
-R impulse responses are collected, and a total of eight types of impulse response data are obtained. Therefore, at the time of reproduction, another set of the configuration shown in FIG. 10 needs to be prepared. Similarly, although not shown, the microphone 103
A center microphone may be added between FL and 103F-R.

FIG. 11 shows an example of the configuration of the reverberation adding apparatus more specifically. The reverberation adding apparatus 1 converts digital audio signals for two channels (1 ch / 2 ch) into AES / EBU (Audio Engineering Society / Eur).
The signal is input from a digital audio input terminal 10 based on the standard of an opean broadcasting (union). Input terminal 10
Is supplied to the input switcher 12 via the digital input unit 11.

The input digital audio signal is
For example, the sampling frequency is 48 kHz and the number of quantization bits is 24 bits. By mounting an option board 50 to be described later on the device 1, the sampling frequency that can be handled can be doubled to 96 kHz. Further, the present invention is not limited to these examples, and can be adapted to digital audio signals having a sampling frequency of 44.1 kHz, for example. In this case,
When the option board 50 is mounted, a signal having a sampling frequency of 88.2 kHz can be handled.

When an analog audio signal is input to the reverberation adding device 1, the analog audio input terminals 13L and 13R are used. L (left) and R
Each of the (right) channel audio signals is input from the corresponding side of the input terminals 13L and 13R, and A
The number of quantization bits is sampled at a sampling frequency of 48 kHz by the / D converter 14 at 24 bits, for example.
It is converted to a digital audio signal. The output of the A / D converter 14 is supplied to the input switcher 12.

The input switcher 12 switches the system of the input audio signal under the control of a controller 40 described later or by a manual switch. The output of the input switcher 12 passes through a path 31 to D
It is supplied to an SP (Digital Signal Processor) 30.

The DSP 30 has a DRAM (Dynamic Random).
Access Memory), and performs various controls on input / output digital audio signals based on a program supplied from a controller 40 described later. The DSP 30
The supplied digital audio signal is supplied to DSPs 32A to 32K for performing a convolution operation of an impulse response based on a predetermined program. Also, DSP
At 30, an initial reflected sound is generated based on the input signal. Further, the DSP 30 is supplied with a convolution operation result of the impulse response from a DSP 34 described later.

Each of the DSPs 32A to 32K cuts out the digital audio signal supplied from the DSP 30 into blocks of a predetermined size, and performs a convolution operation based on previously supplied impulse response data. DSP3
Each of 2A to 32K has a DRAM having a capacity corresponding to the number of samples to be processed. In this example, DSP32A ~
32H is one each, DSP32I is two, DS
Each of P32J and 32K has a capacity of 16M bits.
Has RAM.

The result of the convolution operation of the impulse response for each block performed by the DSPs 32A to 32K is as follows:
The signals are added by the adder 33, and the DSP 30
Supplied to In the DSP 34, an overflow of the addition result is detected, and for example, the data in which the overflow has occurred is fixed to a predetermined value.

The DSP 30 mixes the input digital audio signal, the above-mentioned initial reflection sound, and the result of the convolution operation of the impulse response supplied via the DSP 34 to generate a reverberation sound for the input digital audio signal. Add and output. The output 35 of the DSP 30 is supplied to the output switcher 18.

The formed reverberation sound and the unprocessed input digital audio signal are also referred to as “wet component” and “dry component”, respectively. DS
In P30, the mixing ratio of the wet component and the dry component can be freely changed for each of the L and R channels. At the same time, the DSP 30 also adjusts the level of the output signal.

Further, the clock FS or 2FS having a frequency corresponding to the sampling frequency of the digital audio signal to be handled is supplied to the DSP 30. D
The signal processing in SP30 is performed based on this clock.

The output switcher 18 switches the output signal system under the control of a controller 40 described later or by a manual switch. The output can be output as digital and analog audio signals. An output terminal 20 according to the AES / EBU standard from the output switcher 18 via the digital output unit 19
, A digital audio signal for two channels is derived. The digital audio signal output from the output switcher 18 is converted by the D / A converter 21 into analog audio signals of the L and R channels. The analog audio signals of the L and R channels are supplied to analog output terminals 22L and 22L, respectively.
Derived to 2R.

In this example, the input terminal 10, the input terminals 13L and 13R, the output terminal 20, and the output terminal 22L
And 22R each use a cannon type having three signal lines of hot, cold and independent ground lines.

Further, the output switcher 18 can be selected so as to bypass the reverberation sound adding process in the device 1 for the input audio signal. When the bypass is selected, the input digital audio signal is supplied directly from the input switcher 12 to the output switcher 18 through the bypass path 17.

On the other hand, the entire reverberation adding apparatus 1 is controlled by the controller 40. Controller 40
Is, for example, a CPU (Central Processing Unit) or RAM
(Random Access Memory), ROM (Read Only Memory),
It comprises a predetermined input / output interface. ROM
For example, an initial program for activating the system and a serial number are stored in advance. RAM is CPU
Is a work memory for operating, and a program is externally loaded, for example.

The controller 40 is connected to the bus 41 in 8-bit parallel, for example. The bus 41 is connected to the DS
P30, 32A to 32H, 34 respectively.
Via the bus 41, the controller 40 and each DSP 30,
Communication is performed with 32A to 32H and 34. With this communication, the controller 40 sends each DSP 30, 32A
To 32H and 34H, the controller 40 and each of the DSPs 30 and 32A
Data and commands are exchanged between 32H and Ｈ32H.

Further, as described above, the input switcher 12 and the output switcher 18 are connected to, for example, the bus 41 (not shown), and
Is controlled by

The controller 40 is connected to a display device 42 composed of, for example, a full-dot LCD (Liquid Crystal Display). A predetermined display is performed on the display device 42 based on the display data generated by the controller 40.

Although not shown, the input section 43 has a plurality of input means, for example, a rotary encoder adapted to input data corresponding to a rotation angle, and a plurality of push switches. By operating these input units, corresponding control signals are supplied from the input unit 43 to the controller 40. Based on this control signal, predetermined programs, parameters, and the like are supplied from the controller 40 to the DSPs 30, 32A to 32H, and 34.

The reverberation adding device 1 includes a CD-ROM (C
An ompact Disc-ROM) drive 44 is provided. CD-R
A CD-ROM 45 is inserted into the OM drive 44, and data and programs are read from the CD-ROM 45. The read data and program are CD-R
The data is supplied from the OM drive 44 to the controller 40.

For example, the impulse response data is recorded on the CD-ROM 45. The impulse response data is read from the CD-ROM 45 and supplied to the controller 40. Then, from the controller 40, DS
This data is supplied to each of P32A to 32K. The DSPs 32A to 32K perform a convolution operation of the impulse response based on the supplied impulse response data.

By recording a large number of impulse response data collected in various environments on the CD-ROM 45, the same reverberation effect as in the environment corresponding to the impulse response to be used can be obtained. Further, a plurality of impulse response data can be used in combination. It is possible to create a space that does not actually exist. Further, the impulse response data can be processed by the reverberation adding device 1. For example, the reverberation time is adjusted by processing the read impulse response data and performing a fade-out process.

As another example, a CD-ROM 45
Alternatively, data obtained by converting impulse response data into frequency element data by Fourier transform may be recorded. The processing in the reverberation adding device 1 can be reduced.

Further, the CD-ROM 45 also stores display data used for displaying on the display section 42 described above.

The reverberation adding apparatus 1 has a MIDI (Musical Instrument Digital Interface) as an external interface.
rface). The MIDI signal supplied from the MIDI input terminal 46 is supplied to the controller 40. Based on the supplied MIDI signal, a predetermined function of the device 1 can be controlled. Further, the controller 40 can generate and output a MIDI signal.
The MIDI signal supplied from the MIDI input terminal 46 can be processed and output. The MIDI signal output from the controller 40 is supplied from a MIDI output terminal 47 to an external device. In addition, the MIDI through terminal 48 is connected to the MIDI input terminal 46.
The signal is output as it is.

The function of the reverberation adding apparatus 1 can be expanded by mounting the option board 50.
As an example of the function expansion, the sampling frequency is 48 kHz.
The digital audio signal of z can handle signals of two systems, that is, signals of four channels in total. Thus, for example, the above-described surround-compatible reverberation sound addition and stereo input / stereo output reverberation sound addition can be performed by one reverberation adding apparatus 1.

As another example of the function expansion, when a digital audio signal for two channels (1ch / 2ch) is handled, a signal whose sampling frequency is twice as high as 96 kHz can be handled.

Digital audio signals for two channels (3 ch / 4 ch) are input from the terminal 15 via the option board 50. This digital audio signal is supplied to the input switcher 12 via the digital input section 16. Further, a digital audio signal for two channels corresponding to the processing in the option board 50 output from the output switcher 18 is led out to the terminal 24 via the digital output unit 23. This digital audio signal is output from the terminal 24 via the option board 50 to the outside.

The option board 50 and the device 1 are
The terminals 51 to 56 and the terminals 15 and 24 are connected to each other. FIG. 12 shows an example of the configuration of the option board 50. This option board 50 is the DSP32A described above.
The convolution operation of the impulse response by .about.32K and the adder 33 can be extended and executed. Therefore, this option board 50 includes DSP32L, 32 similar to DSP32A to 32K described above.
M and DSPs 60A to 60L are provided, and an adder 61 and a DSP 62 corresponding to the above-described DSP 34 are provided.
Are provided.

The bus 41 ′ on the board 50 is connected to the bus 41 of the device 1 via the terminal 56. Each DSP 32L, 32M, and DSPs 60A-L on board 50 are
Communication with the controller 40 can be performed via the bus 41 '.

Each of the DSPs 32L and 32M has eight 16-Mbit DRAMs, and performs a convolution operation together with the DSPs 32A to 32K. An input digital audio signal is output from the DSP 30,
32L and 32M, respectively. DS
The results of the convolution operation by P32L and 32M are supplied to the adder 33 via terminals 54 and 55, respectively, and are added together with the operation results of the other DSPs 32A to 32K.

On the other hand, the DSPs 60A to 60M perform processing in parallel with, for example, the above-mentioned DSPs 32A to 32M. An input digital audio signal is output from the DSP 30,
The signals are distributed to the DSPs 60A to 60M via the terminal 51.

For example, when the processing of four channels from 1ch to 4ch is performed by mounting the option board 50, the convolution calculation of 1ch and 2ch is performed by the DSPs 32A to 32M, and the DSPs 60A to 60C are processed.
The convolution operation of 3ch and 4ch is performed by 0M. When a digital audio signal having a sampling frequency of 96 kHz is handled, for example, DSPs to which blocks having the same number of samples are supplied, that is, DSPs 32A and 60A, DSPs 32B and 60B,. By performing the convolution operation in parallel, it is possible to cope with the processing at double speed.

The convolution operation results in the DSPs 60A to 60M are supplied to the adders 61 and added. The addition result is supplied to the DSP 62, subjected to overflow processing similarly to the above-described DSP 34, and
It is supplied to SP30. Then, in the DSP 30,
If necessary, adjust the ratio of the dry component and the wet component, and adjust the mixing ratio with the signals of other channels,
The output is supplied to the output switcher 18.

Note that the option board 50 includes AES
An input terminal 63 and an output terminal 64 for a digital audio signal based on the / EBU standard are provided. A signal for two channels (3 ch / 4 ch) is input to the input terminal 63, and the input signal is supplied to the input switcher 12 via the terminal 15. Similarly, the two channels (3c) output from the output switcher 18
The output signal for (h / 4ch) is supplied to the board 50 via the terminal 24 and is led out to the output terminal 64. In this example, the terminals 63 and 64 are of a cannon type.

FIG. 13 shows an example of the front panel 200 of the reverberation adding apparatus 1. At four corners of the front panel 200, mounting holes are provided so that the device 1 can be mounted on a rack. Panel 200
A power switch 201 is provided on the left side of the CD-ROM, and a CD-ROM insertion unit 202 for mounting the CD-ROM 45 in the CD-ROM drive 44 is provided below the power switch 201. By operating the switch 205, the insertion of the CD-ROM 45 into the CD-ROM
02 can be taken out of the CD-ROM 45.

The display unit 20 is provided substantially at the center of the panel 200.
3 are provided. The display unit 203 uses the above-described LCD 42
It corresponds to. On the right side of the display unit 203, a rotary encoder 204 is provided. The display unit 203
Function keys 206, 207, 208
And 209 are provided. By using the rotary encoder 204 and the function keys 206 to 209, it is possible to select a function of the apparatus 1 and input data.

The display unit 203 performs various displays according to the selected function and the like. In this example, a parameter is displayed when a predetermined reverberation type is selected,
In the display area 210 in the display unit 203, parameters designated for the selected reverberation are sensuously displayed, and in the display area 211, parameter names and parameter values are displayed.

The display in the display area 211 is performed by the function switches 206 to 20 arranged below the display section 203.
9 respectively. For example, by pressing any of the function keys 206 to 209, the parameter displayed immediately above the pressed key is selected. Then, when the rotary encoder 204 is rotated, its parameters are changed. Further, for example, another page can be displayed on the display unit 203 by a predetermined operation. On another page, another parameter value can be changed.

On the other hand, in this embodiment, a ripple corresponding to the currently set parameter value is displayed in the display area 210, and the effect of the reverberant sound (spread of sound) by the parameter value is sensed. It is made to be able to grasp. FIG. 10 and FIG.
An example of the display is shown. As the reverberation time is changed from a short value to a long value, the wave numbers of the ripples are increased as shown in FIGS. 14A to 14H and FIGS. 15A to 15H.

In this example, the ripple has a 16-stage display corresponding to the values of the reverberation time from the minimum value to the maximum value in a stepwise manner. This 16-level display is relative to the reverberation time. The display data for the ripple display is CD-R.
It is stored in the OM45. And, for example, this device 1
Is read from the CD-ROM 45 in advance at the time of startup, and is stored in the RAM of the controller 40. The present invention is not limited to this, and may be stored in the ROM of the controller 40 in advance. When the parameter of the reverberation time is determined, the display of the ripples is maintained at the display at that time.

By performing such display, a visual impression can be given to the user. The user can intuitively grasp the effect of reverberation. That is, the user can visually grasp the spread of the reverberation sound from the ripples.

In this example, the ripples are displayed so as to spread from the lower left to the upper right of the display area 210 in this example, but the present invention is not limited to this example. FIG.
Shows another example of the display of ripples on the display area 210. The center point of the ripple and the direction in which the ripple spreads can be set arbitrarily. For example, the left end can be the center of the ripple (FIG. 16A). Also, the center of the display area 210 can be set as the center of the ripple (FIG. 16B). Furthermore, a cross section of a ripple may be displayed (FIG. 16).
C). Also, the shape of the ripples can be changed according to the type of reverberation sound selected. Further, in this example,
Although the display of the ripples is performed in a fixed manner, an animation display can be made by preparing a plurality of display data for one-step parameters and continuously switching and displaying these.

Next, the DSPs 32A to 32M and the DSP 60
The convolution operation of the impulse response performed in A to 60M will be described. Here, in order to avoid complexity, the DSP 32A is used without using the option board 50.
An operation performed only at ~ 32K will be described.

FIG. 17 schematically shows the processing in each of the DSPs 32A to 32K. The impulse response data is
Under the control of the controller 40, for example, a CD-ROM
45, are supplied to the DSPs 32A to 32K in advance, and are provided to the DSPs 32A to 32K.
Stored in RAM. And each DSP32A-32K
, The impulse response data is divided at predetermined intervals on the time axis, corresponding to the processing block size determined for each.

Here, each of the DSPs 32A to 32K is a DSP
32, and the unit of the impulse response processed by the DSP 32 is N. For example, in this example, DSP3
In 2A, since convolution operation of 128-point impulse response data is performed, N = 128. In the following description, one word corresponds to one sampling data of a digital audio signal. Therefore, one word has a time interval of (1 / sampling frequency) on the time axis, and has a quantization bit number (24 bits) as digital data.

The input data supplied to the DSP 32 is N
It is cut out into block data consisting of words. Therefore, the first N words of time are spent on data input. The input N-word data is stored in the DSP32
Is stored in the DRAM. Then, the convolution operation of the impulse response to the stored N words of input data is performed in the next N words. When all the calculations are completed, the calculation results for N words are output.
Therefore, in the operation of N words, a delay of 2N words occurs for the input / output of data.

FIG. 18 shows the processing in the DSP 32 in more detail. DSP32 is a well-known technology.
The convolution operation of the impulse response is performed using the overlap save method in the cyclic convolution.

That is, as shown in FIG. 18, the n-th block 80B and the immediately preceding (n-1) -th block 80A supplied every N words according to the time axis.
And a DFT (Discrete Fourier Transform) is performed on the data and the data on the time axis is converted to the real part 8 of (N + 1) words.
1A and an imaginary part 81B of (N-1) words.

On the other hand, the impulse response data 82 is subjected to DFT in advance for the N words of the real data 82A and the zero data 82B, and the real part 83 of (N + 1) words is obtained.
It is converted into frequency element data 83 comprising A and an imaginary part 83B of (N-1) words.

Frequency element data 81 based on input data
And corresponding frequency elements of the frequency element data 83 based on the impulse response are multiplied by each other, and a filter process (convolution) of adding equal frequency components to each other is performed on the multiplication result. As a result of this operation, (N +
1) Frequency element data 84 consisting of the real part 84A of the word and the imaginary part 84B of the (N-1) word is obtained. The frequency element data 84 is subjected to IDFT, which is the inverse processing of DFT, to obtain data 86 on the time axis consisting of 2N words.

The result of the IDFT is shown in FIG.
As shown at 86 and 87, 2N words are obtained at N word intervals. In each of the data 85, 86, and 87, the first-half N-word data 85A, 86A, and 87
A is discarded, and output data is obtained, such as the (n-1) th block, the nth block, the (n + 1) th block, and so on. The n-th output data is
Two blocks for the corresponding n-th input data,
I'm late.

By increasing the block size and performing a convolution operation of more impulse response data in one process, a long reverberation time can be obtained. However, as described above, there is a delay of two blocks before the input block is output. Therefore, if one block is increased, the delay time until the reverberation processing component is output increases, Not practical. Therefore, in this embodiment, processing for obtaining a desired reverberation time is performed by:
This is performed in parallel for each of a plurality of blocks divided into a predetermined number of points (number of words).

FIG. 19 and FIG. 20 show the convolution operation processing divided into a plurality of blocks according to this embodiment. For example 2 ¹⁸ words Consider the case of performing the convolution calculation of (256k words). In this case, the digital audio signal is convolved with impulse response data of 256 k words (256 k points).
Approximately 5.3 seconds when the sampling frequency is 48 kHz
c, a reverberation time of approximately 5.9 sec is obtained when the sampling frequency is 44.1 kHz.

As shown in FIG.
A 6k word is divided into two parts, and a part located ahead on the time axis is further divided into two parts. As described above, the side located forward on the time axis is sequentially divided into two. And
Of the two divisions, each of the rear sides on the time axis is further divided into two to form two blocks of the same size.

FIG. 20 shows an enlarged portion A of the first 8 k words in FIG. This part A is also divided into two parts in the same manner, but for the first 256 words,
Two blocks of 128 words are formed.
The impulse response is convolved for the block. Therefore, the reverberation component is output after being delayed by the first 256 words. However, for example, when the sampling frequency is 48 kHz, this is a delay of only 5 msec, and there is no problem in terms of adding reverberation.

[0132] In this way, the entire 2 ¹⁸ words (256k
In this example of the word), 2 ⁷ words (128 words),
2 ⁸ words (256 words), 2 ⁹ words (512 words), 2 ¹⁰ words (1k word) 2 ¹¹ words (2k word) 2 ¹² words (4k words), 2 ¹³ words (8k
Word), 2 ¹⁴ words (16 k words), 2 ¹⁵ words (32 k words) and 2 ¹⁶ words (64 k words)
Of 2 ⁿ words each having a size of 2 ⁿ
It is formed block by block.

In the DSPs 32A to 32K, processing is performed on sets of the same block size. That is, as shown in FIG. 19 and FIG.
Input data supplied to A to 32K is DSP3
In each of 2A to 32K, 12 for DSP32A
8 words, 256 words for DSP32B, DSP32C
512 words for DSP, 1k words for DSP32D, DSP3
2k words for 2E, 4k words for DSP32F, DSP
8k words for 32G, 16k words for DSP32H, DS
32k words for P32I, 64 for DSP32J and 32K
Each is cut out into k words.

In each of the processes from 128 words to 32k words, the convolution process for two blocks having the same block size is performed by one DSP in a time-division manner.

That is, in each of the DSPs 32A to 32K, a convolution operation is performed on the cut-out block data using the corresponding impulse response data. The latter half of the set of the same block size is output after being delayed by one block after processing.
As a result, in each of the DSPs 32A to 32K, two blocks of the same size are continuously output. D
The reverberation data 88 is generated by adding the outputs of the SPs 32A to 32K by the adder 33.

The data continuously supplied to each of the DSPs 32A to 32K is
It is well known that reverberation can be added to continuously supplied data by performing processing in each cycle of A to 32K and adding the results.

FIG. 21 shows a convolution filter 7 for performing a convolution operation in each of the DSPs 32A to 32K.
0 shows an example of the configuration. The convolution filter 70 is realized based on a predetermined program supplied from the controller 40 to the DSPs 32A to 32K, for example. A digital audio signal is input from a terminal 71 and supplied to a DFT circuit 72. The digital audio signal is converted by the DFT circuit 72 from data on the time axis into frequency element data. The output of the DFT circuit 72 is
The signal is supplied to the multiplier 74 and also to the delay circuit 73.

The delay circuit 73 has a delay of N words. That is, each of the DSPs 32A to 32K has N =
128,256,512,1k, 2k, 4k, 8k, 1
6k, 32k and 64k with corresponding delays. The data delayed by the delay circuit 73 is
6.

The multiplier 74 is supplied with a filter coefficient A as impulse response data subjected to DFT from a terminal 75. In the multiplier 74, the output of the DFT circuit 72 and the corresponding frequency element of the filter coefficient A are multiplied. On the other hand, a similar process is performed in the multiplier 76. That is, the filter coefficient B which is the DFT impulse response data is supplied from the terminal 77, and the output from the delay circuit 73 and the corresponding frequency element of the filter coefficient B are multiplied.

The multiplication results of multipliers 74 and 76 are added by adder 78. The addition result is supplied to the IDFT circuit 79, where the frequency element data is converted into data on the time axis, and output from the terminal 80.

As described above, the convolution filter 70 performs a convolution operation using two blocks of data of input data and N words, that is, input data delayed by one block. Is output. As already described with reference to FIG. 14, the first half of the output two-word data is discarded.

FIG. 22 is based on the configuration of FIG.
The processing of the convolution filter 70 is shown corresponding to the time axis. The input data is shown on the left end side of FIG. 22, and the output data is shown on the right end side. FIG. 22 shows the passage of time from the upper side to the lower side as a whole. That is, although a plurality of filters 70 are illustrated as being present, these indicate processing of one filter at different timings. In this way, D
The result of the FT is delayed by the delay circuit 73 and used for the filtering process at the next timing. Therefore, output data delayed by two blocks with respect to the input data is continuously output.

FIG. 23 is a functional block diagram schematically showing the parallel processing of the DSPs 32A to 32K. Input data is D
It is supplied in parallel to each of the SPs 32A to 32K. The DSPs 32A to 32K have N = 128,
N = 256, N = 512, N = 1k, N = 2k, N = 4
k, N = 8k, N = 16k, N = 32k and N = 64
Performs convolution of k points. The operation result is delayed by 2N words from each of the DSPs 32A to 32K and supplied to the adder 22.

For example, the input data supplied to the DSP 32A is cut out into blocks each having N = 128 words, a convolution process is performed on the cut out blocks, and the result of the operation is delayed by 2N words with respect to the input timing. Is output. Then, the next block of N words is fetched, and the same processing is repeated. DSP32B
A similar process is performed in each of ~ 32K.

In the above description, the impulse response collecting device 97 and the reverberation adding device 1 are described as being separate devices, but this is not limited to this example. That is, the reverberation adding apparatus 1 is provided with a measurement signal generator 90 for generating a TSP signal, a synchronous adder 94, and an impulse response converter 95. Needless to say, these can be constituted by a CPU and some peripheral components. DSP 30 or D originally included in reverberation adding device 1
It is also possible to use SP34 or the like. In this way, by providing the reverberation adding device 1 with a function of collecting impulse responses, a user-specific sound effect can be obtained.

Further, in the above description, the convolution processing of the impulse response is performed by hardware such as the DSPs 32A to 32K. However, this is not limited to this example, and can be performed by software processing. Similarly, DSP
The processes of 30 and 34 can also be performed by software.

Further, in the above description, the present invention has been described as supporting the 5.1-channel or 4-channel surround system. However, the present invention is not limited to this example. By installing a microphone at a position corresponding to the reproduction method when collecting impulse responses, the present invention, for example, arranges two speakers in each of the L and R channels on the front, and arranges two speakers in the L and R channels in the rear.
The present invention can be applied to a system in which a book speaker is arranged.

[0148]

As described above, according to the embodiment of the present invention, the reverberation sound is generated by convoluting the entire impulse response actually obtained by the measurement in the real space. Thus, there is an effect that a high-quality processing result can be obtained. That is, according to this embodiment, there is an effect that the pitch of the reverberant sound is equal to the pitch of the input sound.

According to one embodiment of the present invention, the impulse response is measured a plurality of times by the TSP signal,
Since the obtained results for a plurality of times are obtained by synchronous addition, the effect of realizing a higher S / N ratio than an actual reverberation sound obtained from, for example, an iron plate echo device or an actual space can be achieved. is there.

Furthermore, according to one embodiment of the present invention,
By processing the impulse response data obtained from the real space, there is an effect that it is possible to adjust the reverberation time, which cannot be performed in the real space or the like.

Furthermore, according to the present invention, since the output of four devices can be controlled by one device, there is an effect that the operation is easy.

Further, according to the present invention, there is an effect that a natural sound image localization can be obtained even during stereo input.

Further, according to the present invention, there is an effect that a sound effect of microphone setting similar to that of recording in an actual hall can be easily obtained.

[Brief description of the drawings]

FIG. 1 is a schematic diagram showing reverberation sound according to an embodiment in comparison with reverberation sound obtained by a conventional recursive filter.

FIG. 2 is a block diagram showing an example of a configuration of an impulse response collection device according to the present invention.

FIG. 3 is a schematic diagram illustrating an example in which an impulse response is collected in a hall.

FIG. 4 is a schematic diagram illustrating an example of an impulse response processing.

FIG. 5 is a schematic diagram illustrating an example of an impulse response processing.

FIG. 6 is a block diagram schematically illustrating an example of a configuration of a reverberation adding device that performs convolution using impulse response data.

FIG. 7 is a schematic diagram illustrating an example of a case where an impulse response is collected so as to correspond to a surround system.

FIG. 8 is a block diagram illustrating an example of a configuration of a reverberation adding device that adds reverberation sound to input data in a surround method.

FIG. 9 is a schematic diagram illustrating an example of a case where an impulse response is collected so as to correspond to a stereo input.

FIG. 10 is a block diagram illustrating an example of a configuration of a reverberation adding device that adds reverberation sound in stereo to data input in stereo.

FIG. 11 is a block diagram more specifically showing an example of the configuration of a reverberation adding apparatus.

FIG. 12 is a block diagram illustrating an example of a configuration of an option board of the reverberation adding device.

FIG. 13 is a schematic diagram illustrating an example of a front panel of the reverberation adding device.

FIG. 14 is a schematic diagram illustrating an example of a ripple displayed in a display area.

FIG. 15 is a schematic diagram illustrating an example of a ripple displayed in a display area.

FIG. 16 is a schematic diagram illustrating another example of a ripple displayed in a display area.

FIG. 17 is a schematic diagram schematically showing processing in each DSP that performs a convolution operation.

FIG. 18 is a schematic diagram illustrating processing in each DSP in more detail;

FIG. 19 is a schematic diagram illustrating a convolution calculation process divided into a plurality of blocks.

FIG. 20 is a schematic diagram illustrating a convolution calculation process divided into a plurality of blocks.

FIG. 21 is a block diagram illustrating an example of a configuration of a convolution filter in each DSP.

FIG. 22 is a schematic diagram illustrating processing of a convolution filter corresponding to a time axis.

FIG. 23 is a schematic diagram illustrating an example in which processing of different N words is performed in parallel.

[Explanation of symbols]

1 ... reverberation adding device, 30 ... DSP, 32A-3
2M ... DSP, 33 ... Adder, 34 ... DS
P, 40: controller, 42: LCD display unit, 43: input unit, 44: CD-ROM drive, 45: CD-ROM, 50: option board, 60A- 60M DSP, 61 Adder, 62 DSP, 90 Measurement signal generator, 94 Synchronous adder, 95 Impulse response converter 95, 96L, 96R , 96F-L, 96F-
R, 96R-L, 96R-R, 96L / FL, 96L
/ FR, 96R / FL, 96R / FR impulse response data, 97 impulse response collection device, 101 hall, 102L, 102R sound source, 103FL, 103FR, 103LR, 103RR
... Microphone, 122 ... Convolution filter

Claims (9)

[Claims] 1. A method of collecting impulse responses used for generating a sound effect by convolving an impulse response, wherein a spatial sound generated based on a measurement signal is spatially arranged. A method for collecting impulse responses, comprising: a measuring step of measuring each of the at least one measuring means; and a converting step of converting each of the measured sounds into an impulse response. 2. A method for collecting an impulse response used for generating a sound effect by convolving an impulse response, comprising the steps of: generating a sound effect based on a measurement signal at each of two points corresponding to a stereo system; A step of measuring sound by each of two or more spatially arranged measuring means; and a step of converting each of the measured sound into an impulse response. How to collect impulse response. 3. The impulse response collecting method according to claim 1, wherein the measurement signal is generated a plurality of times, and a measurement result of the measurement signal generated a plurality of times is synchronously added. A synchronous addition step of performing the conversion into the impulse response based on a result of the addition in the synchronous addition step. 4. The method for collecting impulse responses according to claim 1, further comprising a step of processing the impulse response converted in the step of converting. Method. 5. A step of measuring sound in a space generated based on a measurement signal by each of four or more spatially arranged measuring means, and impulse each of the measured sound. A computer-readable recording medium on which is recorded impulse response data obtained from the step of converting into a response. 6. A measurement step of measuring spatial sound generated based on a measurement signal at each of two places corresponding to a stereo system with each of two or more spatially arranged measuring means; A conversion step of converting each of the measured sounds into an impulse response. A computer-readable recording medium having recorded thereon impulse response data. 7. The recording medium according to claim 5, wherein the measurement result obtained by the measurement signal generated a plurality of times is synchronously added to the impulse response using an addition result. A computer-readable recording medium characterized by performing the following. 8. A sound effect adding apparatus for adding a sound effect by convolving impulse response data with an input digital audio signal, wherein a plurality of impulse response data respectively corresponding to a plurality of places in the same space are recorded. Reproducing means for reproducing the plurality of impulse response data from the recording medium; input means for inputting a digital audio signal; and the plurality of impulse response data reproduced by the reproducing means. A sound effect adding device comprising a plurality of convolution means for convolving digital audio signals, respectively. 9. The sound effect adding apparatus according to claim 8, wherein the digital audio signal is input in stereo,
A sound effect wherein the convolution is performed for each of the stereo inputs by the plurality of convolution means, and each of the outputs of the plurality of convolution means is mixed corresponding to the stereo input. Additional equipment.