JP2000099066A - Display method and effect sound adding device - Google Patents

Abstract

PROBLEM TO BE SOLVED: To easily comprehend the degree of the spread of sounds in the case of adding reverberation sounds. SOLUTION: In order to display the parameters of reverberation sounds, the device displays ripples corresponding to the length of reverberation time that is set. As an example, as the reverberation time varies from a short value to a long value, the intervals of the ripples are stepwise varied as shown in Figures A to H and at the same time the number of ripple waves is increased. Thus, reverberation time is visually grasped and the setting of the time is properly matched with the feeling.

Description

DETAILED DESCRIPTION OF THE INVENTION

[0001]

BACKGROUND OF THE INVENTION 1. Field of the Invention The present invention relates to a display method and a sound effect adding device that allow a user to easily grasp the parameters of a reverberant sound when adding the reverberant sound to an audio signal.

[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.

[0004]

In the reverberation sound adding apparatus described above, various parameters of the reverberation sound to be added can be changed to adjust the degree of effect. More important of the many parameters,
There is a parameter that indicates the extent of the sound. For example, the extent of the sound can be changed by changing the reverberation time.

Conventionally, the degree of spread of sound has been represented by the display of reverberation time by a mathematical method such as a numerical value or a bar graph. However, there is a problem that the display of numerical values and bar graphs is inorganic, and it is difficult to intuitively grasp the extent of the sound.

In addition, there has been a problem in the past that there was no expression method capable of visually detecting the degree of spread of the sound in an easily understandable manner.

SUMMARY OF THE INVENTION Accordingly, it is an object of the present invention to provide a display method and a sound effect adding device capable of easily grasping the extent of a sound.

[0008]

According to the present invention, there is provided a display method for displaying characteristics of a sound effect added to an audio signal in order to solve the above-mentioned problem. This is a display method characterized by displaying a ripple according to the characteristic of the image.

Further, the present invention provides a sound effect adding apparatus for adding a sound effect to an audio signal. Display means for displaying ripples according to characteristics.

As described above, according to the present invention, a ripple corresponding to the characteristics of the reverberant sound added to the audio signal is displayed, so that the characteristics of the reverberant sound can be intuitively grasped.

[0011]

DETAILED DESCRIPTION OF THE PREFERRED EMBODIMENTS One embodiment of the present invention will be described below. The sound effect adding device according to this embodiment is a reverberation sound adding device that adds a reverberant sound to an original sound that is an input digital audio signal,
The input digital audio signal is convolved with the impulse response data obtained by collecting the reverberation of an actual hall or the like to obtain a reverberation sound to be added. Then, a ripple display according to the length of the reverberant sound is made on the display unit. This makes it easy to intuitively grasp the length of the set reverberation sound, that is, the extent of the sound.

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

FIG. 2 shows an example of the configuration of the impulse response collecting device 97 according to this embodiment. In this example, the impulse response of the iron plate echo device 92 is measured. The 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 of the N times are synchronized in the synchronous adder 94 at the timing of signal generation, and are synchronously added. 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 (10 logN) d
B is improved. For example, the S / N ratio is 12d at N = 16.
B is improved.

The synchronously added signal is supplied to the impulse response converter 95 for each of the L and R channels. 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 an appropriate recording medium such as an MO. 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 original sound in the original impulse response in terms of the number of samples. In this way, the fade-out is performed after the position of the original sound. This is because if the fade-out starts at the same position as the original sound, that is, at the time of n = 0, the level of the original sound itself also decreases.

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

Further, by fading out, the number of points of the impulse response data can be adjusted so as to be adapted to the processing capability of a reverberation sound adding apparatus for actually adding a reverberation sound to an audio signal using this 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 becomes out of the point at the point of 256 k points.

[0].

As the processing 111, level adjustment and the like are performed in addition to the above. 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.

The processing of the impulse response data can also be performed by a reverberation sound adding device that adds reverberation sounds using the impulse response data reproduced from the CD-ROM 45, as described later. By performing, for example, the above-described fade-out processing in the reverberation sound adding device, the length of the reverberation sound added to the audio signal can be changed.

FIG. 6 schematically shows an example of the configuration of a reverberation sound 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 is delayed by a predelay 121 to be 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.

FIG. 7 shows an example of the configuration of the reverberation sound 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 / EBU).
The signal is input from a digital audio input terminal 10 based on the standard of the European Broadcasting Union. Input terminal 1
The digital audio signal supplied from 0 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 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.

A 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 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 adding process in the apparatus 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 sound 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.

As described above, the input switcher 12 and the output switcher 18 are connected to, for example, a bus 41 (not shown), and a controller 40.
Is controlled by

A display device 42 composed of, for example, a full-dot LCD (Liquid Crystal Display) is connected to the controller 40. 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 apparatus 1 includes a CD-ROM
A (Compact Disc-ROM) drive 44 is provided. CD-
A CD-ROM 45 is inserted into the ROM 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 CD-ROM 45 records impulse response data. 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 is transferred to the reverberation adding device 1.
Can be processed. 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 apparatus 1 can be reduced.

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

The reverberation sound adding apparatus 1 has a MIDI (Musical Instrument Digital In) as an external interface.
terface). 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. Controller 40
The MIDI signal output from the MIDI output terminal 47
Is supplied to external devices from The MIDI through terminal 48 is connected to the MIDI input terminal 46.
The DI 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 extension, a sampling frequency of 48
kHz digital audio signal, two more systems
Be able to handle. 2 channels (3ch
/ 4ch) digital audio signal is 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
The signal is output from the terminal 24 to the outside via the option board 50.

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.

The option board 50 and this device 1 are
The terminals 51 to 56 and the terminals 15 and 24 are connected to each other. FIG. 8 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 operation 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.

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. 9 shows an example of a front panel 200 of the reverberation sound 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.

At the approximate center of panel 200, display unit 20
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 of 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 the present invention, the display area 21
For 0, a ripple corresponding to the length of the currently set reverberation time is displayed, so that the effect of the reverberation sound (sound spreading) according to the setting can be intuitively grasped. 10 and 11 show examples of the display in the display area 210. FIG. For example, the reverberation time is selected as a parameter based on a predetermined operation of the function key, and the setting of the reverberation time is changed by the rotary encoder 204. As an example, as the reverberation time is changed from a short value to a long value, FIGS. 10A to 10H and FIG.
As shown in FIGS. 1A to 11H, the interval between the ripples is changed, and the wave number of the ripples is increased.

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 parameters of the reverberation time are determined, the display of the ripple is fixed to the display at that time.

By performing such a 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.

Here, the display of the ripples is displayed so as to extend 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.
2 shows another example of the display of the ripples on the display area 210. The center point of the ripple and the direction in which the ripple spreads can be set arbitrarily, and for example, the left end can be the center of the ripple (FIG. 12A). Further, the center of the display area 210 can be set as the center of the ripple (FIG. 12B). Furthermore, a cross section of a ripple may be displayed (FIG. 1).
2C). Also, the shape of the ripples can be changed according to the type of reverberation sound selected.

Further, in this example, the display of the ripples is performed in a fixed manner. However, a plurality of display data are prepared for one-step parameters, and these are continuously switched and displayed. Animation display can also be used. It is displayed as if ripples spread on the water surface.

The reverberation time can be changed as shown in FIGS.
Can be performed by the method described with reference to FIG. That is, for example, the selected impulse response data is multiplied by an exponential function of attenuation. The reverberation time can be changed by changing the parameters of the exponential function. By rotating the rotary encoder 204, this parameter can be changed.

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. 13 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. 14 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. 14, 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 with respect to the N-word real data 82A and the zero data 82B, and the (N + 1) -word real part 83
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. 15 and FIG. 16 show a convolution operation process 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. 16 shows an enlarged portion A of the top 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.

[0091] 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 having the same block size. That is, as shown in FIG. 15 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 extracted 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.

It should be noted that 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. 17 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, which is the DFT impulse response data, 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 the convolution operation using the data of two blocks of the input data and N words, that is, the 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. 18 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 of FIG. 18 and the output data is shown on the right end. FIG. 18 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. 19 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 collection device 97 and the reverberation sound adding device 1 are described as separate devices, but the present invention is not limited to this example. That is,
The reverberant sound 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. DSP3 originally provided in the reverberation adding apparatus 1
It is also possible to use 0 or DSP34. In this way, by providing the reverberant sound adding apparatus 1 with a function of collecting impulse responses, a user-specific sound effect can be obtained.

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 the processing 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 being applied to a reverberation adding apparatus, but the present invention is not limited to this example. For example, the ripple display according to the present invention can be applied to another sound effect adding device that adds a reverberant sound incidentally. That is, the present invention is applicable to a case where reverberation is added together with other effects. Furthermore, the display according to the present invention can be applied, for example, even when the reverberation sound is added by software on a computer.

[0107]

As described above, according to the present invention, a ripple corresponding to the set reverberation time is displayed on the display unit, so that the extent of the sound due to the reverberation can be visually expressed. effective.

Further, according to this embodiment, a ripple corresponding to the set reverberation time is displayed on the display unit, and the extent of the sound caused by the reverberation is visually represented. There is an effect that it can be performed in accordance with the degree of visual spread.

[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 apparatus that performs convolution using impulse response data.

FIG. 7 is a block diagram more specifically showing an example of the configuration of the reverberation sound adding apparatus.

FIG. 8 is a block diagram showing an example of a configuration of an option board of the reverberation adding apparatus.

FIG. 9 is a schematic diagram illustrating an example of a front panel of the reverberation sound adding apparatus.

FIG. 10 is a schematic diagram illustrating an example of a ripple displayed in a display area.

FIG. 11 is a schematic diagram illustrating an example of a ripple displayed in a display area.

FIG. 12 is a schematic diagram illustrating another example of a ripple displayed in a display area.

FIG. 13 is a schematic diagram schematically showing processing in each DSP that performs a convolution operation.

FIG. 14 is a schematic diagram illustrating processing in each DSP in more detail;

FIG. 15 is a schematic diagram illustrating a convolution calculation process divided into a plurality of blocks.

FIG. 16 is a schematic diagram illustrating a convolution calculation process divided into a plurality of blocks.

FIG. 17 is a block diagram illustrating an example of a configuration of a convolution filter in each DSP.

FIG. 18 is a schematic diagram illustrating processing of a convolution filter corresponding to a time axis.

FIG. 19 is a schematic diagram illustrating an example in which processing of different N words is performed in parallel.

[Explanation of symbols]

1 ... Reverberation device, 30 ... DSP, 32A ~
32M ... DSP, 33 ... Adder, 34 ... D
SP, 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 ... Impulse response data, 97 ... .Impulse response collection device, 122...
Convolution filter, 204 ... Rotary encoder, 210 ... Display unit displaying ripples

 ────────────────────────────────────────────────── ─── Continued on the front page (72) Inventor Shinichi Iriya 6-35, Kita-Shinagawa, Shinagawa-ku, Tokyo Sony Corporation F-term (reference) 2G064 AA05 AA11 AB01 AB02 AB12 AB17 BA02 BD02 CC02 CC26 5D108 AA08 AB08 AB09 AD05

Claims (6)

[Claims] 1. A display method for displaying characteristics of a sound effect added to an audio signal, wherein a ripple corresponding to characteristics of a reverberant sound added to the audio signal is displayed. Display method. 2. The display method according to claim 1, wherein the characteristic of the reverberation sound is a reverberation time. 3. The display method according to claim 1, further comprising a changing step of changing a characteristic of the reverberant sound, wherein the display of the ripples is accompanied by a change of the characteristic of the reverberant sound by the changing step. A display method characterized by being changed. 4. A sound effect adding apparatus for adding a sound effect to an audio signal, comprising: a reverberation sound adding unit for adding a reverberation sound to the audio signal; and a characteristic of the reverberation sound added by the reverberation sound adding unit. Display means for displaying a corresponding ripple. 5. The sound effect adding device according to claim 4, wherein the characteristic of the reverberation sound is a reverberation time. 6. The sound effect adding apparatus according to claim 4, further comprising a changing unit for changing a characteristic of the reverberant sound, wherein the display unit is configured to change a characteristic of the reverberant sound by the changing unit. A sound effect adding device, wherein the display of the ripples is changed.