JPH02293795A - Effect sound controller - Google Patents

Abstract

PURPOSE:To allow the dynamic control of the effect characteristics of musical tones by generating an effect sound signal from an effect sound generating means according to performance parameters. CONSTITUTION:Direct digital musical tones are outputted from a sound source 14 via a CPU 11 in accordance with the performance parameters, such as note number PT indicating the pitch from an external controller and velocity VL indicating performance touch, the information in the set change of the performance parameters via an operation panel 19, etc., and are mixed by a 1st mixer 15 of a reverberation effect generating means SE of an effect sound generating means, from which the overall performance parameters are supplied to a reverberation sound generating section 16. The effect sound information from the generating section 16 and the direct sound information from the sound source 14 are mixed by a mixer 17 from which the mixture is outputted. The dynamic control of the effect characteristics of the musical tones corresponding to the set change as well is executed by this constitution.

Description

DETAILED DESCRIPTION OF THE INVENTION [Technical Field of the Invention] The present invention relates to a sound effect control device for musical instrument performance, and more particularly to a technique for controlling how effects are applied in real time.

[Prior art and its problems] Digitalization of sound effectors is progressing in the field of electronic musical instruments. For example, a digitized reverberation adding device (reverb effector) models a real or virtual sound field space (hall, club, etc.) to determine the relationship between the reflected sound and the direct sound (size, shape of the sound field space, etc.).
Determined by reflection coefficient etc. For example, the delay time of the first reflected sound from the direct sound is determined by the difference between the length of the shortest path from the sound source to the listener via the surrounding walls and the length of the messenger path of the direct sound. Its spectral characteristics are determined by the reflection characteristics of walls, the propagation characteristics of air (or other sound propagation media), etc. ), and the relationship is realized using digital circuits and digital signal processing equipment. In the case of a reverberation adding device for a digital electronic musical instrument, the direct sound is provided by an output signal (digital tone signal) of an electronic sound source. The reverberation adding device receives this digital musical sound signal and performs digital calculation to generate a sound effect component signal representing initial and multiple reflected sounds. If the sound effect component signal generated in this way and the output signal of the sound source are emitted through a sound system, a musical sound with reverberation will be heard.

By the way, regarding this type of reverberation adding device, by setting the parameters of the reverberation characteristics on the setting panel (
This is often done as part of setting the characteristics of each sound corresponding to each timbre number), allowing the reverberation adding device to have desired reverberation characteristics. Additionally, in many models, the volume balance between reverberation sound (sound effect component) and direct sound can also be set variably on the settings panel. In extreme cases, if the relative volume of the reverberant sound is set to zero, only the direct sound will be emitted.

However, in current reverberation adding devices, once the reverberation characteristic parameters are set on the setting panel, their status does not change and remains fixed during the performance. As a result, the reverberation during performance becomes monotonous, which limits musical expression.

The problems of the prior art have been shown above by taking the reverberation adding device as an example, but similar problems exist with other effect generating devices.

[Objective of the Invention 1 Therefore, the object of the present invention is to dynamically control the characteristics of the musical sound effect (for example, the degree of the effect sound component signal added to the musical sound signal as the direct sound output from the sound source) according to the performance. It is an object of the present invention to provide a possible sound effect control device.

[Structure, operation, and development of the invention] According to the present invention, in order to achieve the above object, input musical tone signals (given as digital data) generated based on performance operations are digitally processed according to internal effect parameters. By this, in a device having a sound effect generating means for generating a sound effect signal, at least one effect parameter used by the sound effect generating means is generated according to a performance parameter (performance information) characterizing a performance operation, A sound effect control device is provided, characterized in that it includes an effect parameter control means for supplying sound to a sound generation means.

According to this configuration, a performance control input device (for example, a keyboard,
The characteristics of the sound effect dynamically change according to the information that characterizes the performance operation, that is, the performance parameters, provided from a wind instrument controller, a string instrument controller, an external performance controller such as MIDI, an external sequencer, or the sequencer of the instrument itself. Therefore, it is possible to overcome the conventional monotony of sound effects during performances, and the degree of freedom in musical expression is increased.

In one configuration, for example, performance parameters for sound effect control include performance touch (press strength or speed for keyboards, press strength or tonguing strength for wind instruments, pluckedness for stringed instruments). String strength, etc.) and pitch (typically note numbers, for example, in the case of a keyboard, pitch information indicating the key pressed, and in the case of a wind instrument, pitch information determined by the form of fingering for a series of pitch switches) etc.) are used.

In addition, as a sound effect generation means, a configuration of effect units such as reverb, echo, delay, chorus, etc. (single effect configuration), or a configuration in which multiple effect generation modules are connected in cascade so that these effect units can be combined, Or even if it is the same effect unit (
), one or more musical sound signal inputs are divided into multiple parts, each signal is processed independently, and multiple systems of sound effect signals are generated and output. output, two-man power, two-output type, etc.), or a matrix configuration or module that combines a cascade configuration and a parallel configuration. It is possible to use a configuration in which the number of outputs increases toward the end.

However, it is not desirable to have a sound effect generating means that can only output a signal that does not include any direct sound (musical sound signal generated by a sound source) in the output sound. For this reason, the sound effect generating means may be comprised of a sound effect component signal generating means and an output means for outputting a sound source output signal as a direct sound together with the sound effect component signal which is the output of this means. A sound effect generating means having such a configuration can also be used in the present invention.

In one configuration example, the sound effect generation means (or the effect component generation means) includes a plurality of coefficient multiplication means for multiplying the signal related to processing by the effect parameter described in "2" as a coefficient, and the effect parameter control means controls the coefficients used by at least one of the coefficient multiplication means in accordance with the performance parameters described above.

The coefficient multiplication means for controlling the characteristics of the sound effects is typically composed of a digital multiplier.

However, the coefficient multiplier that adjusts the volume balance (amplitude ratio) between the direct sound signal and the effect component signal is not limited to this, and can be an analog multiplier (for example, a voltage-controlled amplifier V
CA) may also be used. For example, the digital output signal of the digital effect component generation means and the digital output signal of the digital sound source may be converted into analog signals, and the amplitude ratio between the analog effect component signal and the analog sound source output signal may be adjusted using a VCA. good.

In addition, when a digital sound source is used as a sound source, the output of the digital sound source can be directly transmitted to the digital sound effect generating means (
However, depending on the situation, it may be convenient to first perform D/A (digital/analog) conversion and then A/D conversion again before inputting it to the sound effect generation means. It's okay.

When employing the volume ratio between the direct sound and the sound effect component signal as a control target (effect parameter) for sound effects using performance parameters, this may be done by adjusting the amplitude ratio on the analog output side, as already described. Alternatively, it may be performed by a digital multiplier provided on the output side of the digital effect component generation means (or sound effect generation means), or by a digital multiplier provided on the input side of the effect component generation means (or sound effect generation means). or analog). However, when the sound effect generation means or the effect component generation means handles fixed-point digital data, it is desirable to perform it on the output side from the viewpoint of numerical precision (accuracy of digital calculation).

When the performance parameter controls something other than the depth (ratio) of the sound effect component, and the sound source is a polyphonic sound source (
In the case of a sound source that generates the musical tones of a polyphonic instrument or a sound source that simultaneously generates the musical tones of multiple monophonic instruments, control may be performed depending on the configuration of the sound effect generating means (or sound effect component generating means). It can be difficult. For example, in the case of a configuration in which polyphonic musical sound signals of different performance sources from polyphonic sound sources are mixed by a mixer to form one total musical sound signal, and this is input to sound effect generation means or effect component generation means, the sound effect Since only the total signal can be processed within the generating means and the effect component generating means, it is impossible to control sound effect characteristics for each performance source. However, even in this case, the depth of the sound effect component or the volume balance between the sound effect component and the direct sound component can be controlled in a manner specific to each performance parameter. That is, in the mixer, the weighting coefficient of the part (multiplying means) that weights the signal for each sound source channel is calculated according to the corresponding performance parameter, that is, the parameter of the performance source that played the sound source channel, This is because it only needs to be supplied.

l3 [Example] Embodiments of the present invention will be described below with reference to the drawings.

First Embodiment> FIG. 1 shows an electronic musical instrument 10 (first embodiment) incorporating the features of the present invention.
Examples) are shown below. This electronic musical instrument 10 generates a reverberant sound signal generated by the reverberant sound generator 16 and a direct musical sound signal generated by the sound source 14 based on pitch information FT and performance touch VL given from an external performance controller (not shown). You can control the volume balance.

In detail, 11 is a system program ROM 12
The RAM 13 is used as a working memory and a memory for tone and effect parameters. This CPU1l includes an external performance controller,
Here, MID I (MIJSIGAL INSTR
UMENTIIIGITAL INTERFACE)
Performance information PER transmitted from the controller is input. As is well known, the MIDI performance information 1APER includes data such as a note number FT indicating pitch and velocity VL indicating a performance touch. For the convenience of the following explanation, assuming a polyphonic mode, the performance information PER is given polyphonically, and the CPU 11 of the receiving electronic musical instrument 10 performs a unique voice assignment for the received sound generation message (note on/off), and performs a unique voice assignment as required. data (timbre parameters) is generated and transferred to the assigned channel of the polyphonic digital sound source 4. Furthermore, in this example,
The CPU II includes a first mixer 15 (MIXl) and a reverberation sound generation section 16 that constitute the digital reverberation effect generation device SE.
, data (effect parameters) necessary for controlling reverberation sound are also transferred to the second mixer l7 (MIX2). In particular, the proportion of reverberant sound is calculated according to at least one of the note number FT and velocity VL included in the note-on message of the performance information PEH.

The tone and effect parameters (sound l5 source l4, parameters used as they are or modified in the reverberation effect generator SE) on the RAM 13 can be set by the user through the display device l8 and the operation panel l9. . For example, the set parameters are grouped by sound tone number, and when a tone change message is given from an external MIDI controller, or in response to a tone selection operation on the operation panel 19 of the main unit, A necessary portion can be sent from the CPU II to the sound source 14 or the reverberation effect generating device SH.

The reverberation effect generator SH has a polyphonic sound source l4.
A plurality of digital musical tone signals D(0) to D(n) are inputted. Specifically, the output of the sound source l4 has the meaning of direct sound, and these digital musical sound signals are input to the first and second mixers l5 and 17. The first mixer l5 forms an input signal for the reverberant sound generator l6, and is configured to form input signals for the plurality of channel-specific musical sound signals D.
(0) to D(n) are mixed to form one total musical tone signal DM, which is supplied to the reverberation sound generating section l6. The reverberation sound generating section l6 forms a signal (reverberation component signal RV) indicating the reflected sound with respect to the direct sound by digital processing, and supplies it to the second mixer 17.

In the second mixer l7, multiple channels of digital musical sound signals D(0) to D are directly provided from the sound source l4.
(n) and the reverberation component signal RV given via the reverberation sound generating section l6 are mixed and outputted to the outside as one digital signal output OUT.

Here, the first and second mixers 15 and 17 will be explained. The configuration of the first mixer l5 is shown in FIG. As shown in the figure, the plurality of channel-specific musical tone signals D(0) to D(n) from the sound source 14 are each weighted in a coefficient multiplier 15-WO-15-Wn for weight adjustment, and then added to an adder. 15-S and input to the reverberation sound generating section 16 shown in FIG. Here, each coefficient multiplier 15-WO to 15-Wn has an independent coefficient (depth parameter) V E L (
0) to VEL(n) are input, and as described later, these coefficients aVEL(0
) to VEL(n) are generated for each channel from the pitch information FT and the velocity VL in the CPU II, and are supplied to these multipliers 15-WO-15-Wn.

FIG. 3 shows an example of the configuration of the second mixer l7, where:
A plurality of channel-specific musical sound signals D (0) to D (n) as direct sounds from the sound source 14 and a reverberation sound generating section l
A signal obtained by weighting the reverberant sound component signal RV from 6 (a signal formed by digitally processing the total signal of musical tone signals weighted by channel in the first mixer 15) with a coefficient RVL in a coefficient multiplier 17-W. RV' is added by the adder 17-S and output to the outside.

As can be seen from FIGS. 2 and 3, the weighting coefficients for each channel VEL(0) to VEL(n
), it is possible to change the volume balance between the direct sound signal D and the reverberant sound component signal RV that are output to the outside. In this embodiment, this volume balance can be dynamically controlled in real time using performance parameters from the performance source, particularly pitch FT and velocity VL (performance touch).

Hereinafter, real-time control of the volume balance between direct sound and reverberant sound components using performance parameters will be explained with reference to FIGS. 4 and 5.

Figure 4 shows the pronunciation message (
This shows the flow of operations of the CPU II executed when a note-on message is given.

Note on message (pitch information FT and velocity V
containing L) is given (4-1), CPUI
l determines the tone channel m to which the note-on message should be assigned (4-2). When the determined assigned channel m is producing sound, that channel m is muted (4-3, 4-4). Does nothing if the channel is already empty. In any case, sound 1i
The 14 assigned channels m are muted.

Next, the CPU II calculates the coefficient VEL(
Calculate m) (4-5). That is, the ratio of the reverberant sound component signal RV(m) to the direct sound signal D(m) is determined. Next, the CPU 11 calculates this calculation coefficient VEL(
m) in the coefficient multiplier 15-Wm of the first mixer l5 (a multiplier that weights the sound source output D(m) of the m-th channel in order to input it to the reverberation sound generator l6) (4-6 ). With the above steps, preparations for channel m in the reverberation generator SE are complete. Finally, CPU11 uses sound source l4
By transferring a sound generation control signal according to FT and VL to channel m, sound source channel m is caused to generate sound (4-7).

As a result, digital musical tone signal D from sound source channel m
(m) rises, and in the first mixer l5, the amplitude is controlled by the coefficient VEL(m) in the coefficient multiplier 15-Wm (total input signal DM of the reverberant sound generating unit l6).
DM(m)) is inputted to the reverberant sound generating section l6, digitally processed, and inputted as the reverberant sound component signal RV(m) to the second mixer 17, where the direct musical sound signal D(m) is added and output. therefore,
A sound whose proportion of reverberant sound components changes according to the pronunciation message will be output.

The form of the operation in steps 4-5 of FIG. 4 may be any suitable. VEL(m
) may be generated, or FT and VL may be converted to VEL(m) using a table. Also, F.T. Any suitable function can be selected for the manner in which VEL(m) changes with respect to the change in VL, that is, the conversion function.

Some examples are shown below.

As a function, choose VEL=l-VL (Figure 5(a)). In this case, the magnitude of the reverberant sound component is determined only by the velocity VL (playing touch), and when the touch is weak (
(VL large), the reverberation becomes small, and when the touch is strong, the reverberation becomes large.

2l VEL=FT (Figure 5(b)) In this case, the size of the reverberation is determined only by the pitch; when the pitch is high (FT large), the reverberation is large, and when the pitch is low, the reverberation is shallow. Become.

VEL= (11+"- (Figure 5 (C)) As in this example, when VEL is a function of both VL and FT, the magnitude of reverberation depends on both pitch and playing touch. For the example function, the reverberation is deepest when the touch is strong and the pitch is high.

Although the description of the first embodiment has been completed above, various modifications and changes can be easily made.

For example, regarding the configuration of mixers 15 and 17, configurations other than those shown in FIGS. 2 and 3 can be adopted. For example,
In the second mixer 17, the direct sound signal D
Coefficient multipliers (17-WO to 17-Wn) that weight (0) to D (n) for each channel are provided, and the coefficients of these coefficient multipliers and the coefficient multipliers that form the input signal to the reverberant sound generating section 16 are provided. Multiplier coefficient VEL of mixer l5 of 1 (0)
〜VEL(n) (typically, α pair (l − α
) may be controlled in a complementary manner, such as the ratio of

Furthermore, in the second mixer l7 in Fig. 3, the direct sound signals D (0) to D (n) and the reverberation component signal RV are added together and outputted to the outside as one total signal OUT. Alternatively, the direct sound signal and the reverberant component signal RV may be output without being mixed (for example, in a case where the direct sound is output from the right speaker and the reverberant sound component is output from the left speaker).
. Furthermore, the first mixer l5 and the second mixer l7 may be constructed from analog circuits. In this case, a D/A converter is attached to the input side of the first analog mixer, and an A/D converter is attached to the output side.
A D/A converter is provided between the digital reverberation sound generator l6 and the analog second mixer. Furthermore, in the first embodiment, since the reverberant sound generator l6 handles only one digital signal, the mixer 15 is required at its input stage. , the corresponding sound source channel signal may be digitally processed in each reverberation sound generation channel. In this case, a coefficient multiplier (digital or analog) for adjusting channel reverberation sound components as shown in FIG. 2 may be provided at the output stage of each reverberation sound generation channel.

Furthermore, the performance controller is not limited to an external performance controller, but may also be a performance controller built into the instrument itself, or is not limited to a real-time performance controller that provides information indicating the performer's on-the-spot performance operations, but is also a medium on which performance information is recorded. It may also be a means (sequencer) for reading performance information from.

Second Embodiment Next, a second embodiment of the present invention will be described. Figure 6 is the second
This is the overall configuration of an electronic musical instrument 10M according to an embodiment. In the second embodiment, the effect generating device SEP is a polyphonic sound source 14.
0 is prepared for each musical tone channel, and effects can be added independently to each channel. Each effect generation unit of the effect generation device SEP SEi has a hardware structure as shown in Fig. 7, and a functional structure as shown in Fig. lθ. Unlike the first embodiment, only reverb is used. In addition, chorus and delay effects can be added in combination according to selection. In FIG. 6, CPUIO
has a built-in system program ROM and data RAM, and responds to sound messages from an external performance MIDI controller to perform voice assignments.
From the information included in that message (in this case, since it is MIDI, pitch information FT and velocity VT), the assigned effect generation unit SE (m
) is calculated and transferred to that unit SE. Thereafter, the CPU LIO gives a sound generation command including required data to the 7-signed sound source channel TG (m) on the sound source 140 to start sound generation. As a result, the sound source channel TG ( m
), a musical tone signal is generated and inputted to the corresponding effect generating unit SE(m), where the effects of reverb, chorus and/or delay are applied to the CP. It is added according to the internal effect parameters updated by U110 at the time of sound generation.

Each effect generating unit SEi is internally divided into two subunits: a right stereo subunit and a left stereo subunit (see Figure lθ), and each subunit is independent or mutually dependent. It is now possible to add effects depending on the format. Among the outputs of each effect generating unit} SEO to SEn, the output of the right subunit is added to the adder L 1 1 1
' and then the left stereo sound system (
(not shown), and the outputs of the left subunit among all the outputs are summed by an adder R112 and then input to the right stereo sound system.

The multiple effect generation units that make up the effect generation device SEP can be multiplexed using time division multiplexing (TDM) using common hardware if the processing speed is high enough, but here they are configured with discrete 9G units. There is.

The hardware configuration of the effect generating unit SEi is shown in FIG.

In the figure, a program memory 2l is a memory that stores a predetermined program, and receives as an address the output of a program counter 22 that is incremented by a high-speed clock signal CK2 supplied from a clock generator (not shown), and executes commands. is supplied to the control circuit 23.

This control circuit 23 decodes the output of the program memory 2l, and controls the timing and execution of data transfer and calculation between each register and memory, which will be described later, and the supply of data to the program counter 22 by the flip-flop 24. The flip-flop 24 is set by an external sampling clock CKI and supplies a set signal F to the control circuit 23, and a reset signal is supplied from the control circuit 23 to the flip-flop 24. Note that the clock signal CK2 supplied to the program counter 22 is given a sufficiently faster clock than the external sampling clock CKI supplied to the flip-flop 24.

The effect parameter memories 25 and 26 store parameters for adding delay, chorus, and reverb effects, constants used in calculations, and part of waveform data, as will be described in detail later.

Register A 27 and register B 28 store data given from effect parameter memories 25 and 26 or each register described later, and supplied to an arithmetic circuit 29 and a multiplication circuit 30 that perform addition and subtraction.

The calculation results of the calculation circuit 29 and the multiplication circuit 30 are applied to the register C31, and the output of the register C31 is supplied to each section via the calculation circuit 29 or the internal bus 32.

The waveform data memory 33 is a memory (delay element) that temporarily stores waveform data of musical tones, and the address register 34
Write and read addresses are supplied by the data register 35, and the write and read data are stored in the data register 35. The waveform data memory 33 is composed of a RAM, etc., and includes shift circuits shown in FIGS. 11 to 13 as delay circuits in the delay effect adding circuit 1, chorus effect adding circuit 2, and reverb effect adding circuit 3 shown in FIG. 10, which will be described later. Functionally corresponds to a register. The data register 35 is bidirectional, and data is transferred via the internal bus 32. The input register 36 has a corresponding sound source channel T.
It is used as a buffer for digital tone signals from G(i). Further, the right output register 38 and the left output register 39 are used for right effect output and left effect output, respectively.

37 is the CPUIO (Figure 1) and the effect generation unit SE
i, and the CPU 110 can change the data in the effect parameter memories 25 and 26 via this interface. Specifically, when changing data, the CPU 110 transfers to the interface 37 a command that includes the data and the address on the effect parameter memories 25, 26 where the data is to be written. On the other hand, ink 1 face 3
7 decodes the command, and the control circuit 23 transfers it to the internal bus 2.
Data is written into the effect parameter memories 25 and 26 at a timing when 3 is not in use.

Next, the internal configuration of the aforementioned timbre parameter memory 2526 will be explained with reference to FIGS. 8 and 9.

FIG. 8 shows the internal configuration of the tone parameter memory 25 in FIG. The delay memory input pointers for the right channel and left channel are placed in the used area of address 3n+1 and 3n+2, respectively.
O I NTL, addresses 3n+3 and 3n+4 are the size of the delay memory used area for the right channel and left channel, DERIAAR and DERIAAL, addresses 3n+5 and 3n+6 are the start address of the delay memory used area for the right channel and left channel, (DER
IAOR and DERIAOL, chorus memory input pointer CPOINT at address 3n+7, address 3n
+8 is the chorus memory usage area CERIA,
Address 3n+9 is the start address CERIAO of the chorus memory use area, address 3n+IO is the reverb memory input pointer RPOINT, address 3n+11 is the size of the reverb memory use area RERIAA, address 3n+12 is the start address RERIAO of the reverb memory use area, addresses 3n+13 and 3n+14. Delay feedback waveform data DRDATAR and DRDATAL for the right channel and left channel, respectively, and processed waveform data WAVER and WA for the right channel and left channel, respectively, at addresses 3n+15 and 3n+16.
VEL, right and left effect output waveform data EWAVER and EWAV at addresses 3n+17 and 3n+18, respectively.
Memorize EL.

FIG. 9 shows the content structure of the timbre parameter memory 26 shown in FIG. 7. Addresses 0 to l1 correspond to the contents of the arbitrary waveform for low frequency oscillation, and LFO use areas 31 and addresses 12 and 13 respectively. The delay for the right channel and the left channel, that is, the 11th delay described later.
DTIMER and DTIMEL corresponding to the contents of the delay time corresponding to the shift registers la and lb in the figure, and addresses l4 and 15, respectively, are filled with delay feedback amounts for the right channel and left channel, that is, a functional block as a delay effect adding circuit described later. The feedback amounts DRPEATR and DRP correspond to the attenuation circuits le and If of FIG.
EATL, address 16 and l7 are the delay effect depth DDEPT for the right channel and left channel, respectively.
HR and DDEPTHL, addresses 18 and 19 respectively indicate the depth of the chorus effect CDEPTH and the delay time of the chorus, that is, the shift register 2a.
Delay time CDTIME corresponding to 2b, address 20 ~
20+m indicates each delay time of the right channel reverb, that is, each delay time corresponding to the intermediate tap of the shift register 3a in FIG. 13, which shows a functional block as a reverb effect adding circuit to be described later.
Each delay time DTIL-DTmL of the left channel reverb is stored in 1+m to 21+2m, and the depth of the reverb effect RDEPTH is stored in address 22+2m, respectively. Here, the actual delay time data DTIME differs from the delay time by the shift register etc. in FIGS. 11 to 13, and is the difference between the addresses on the waveform data memory 33, that is, the address where the current waveform is written and the waveform written in the past. The size of the memory (for example, delay memory) used in one loop (DERIA)
It has the value obtained by subtracting the original delay time from A).

FIG. 10 is a functional block diagram of the effect generation unit SEi. In other words, the effect generating unit SEi has the configuration shown in FIG. 7 and executes the function shown in FIG. 10 in a time-division manner for each sampling. That is, in the same figure, the effect generating unit is a two-man power unit connected in cascade.
It has a two-output type delay effect adding circuit 1, chorus effect adding circuit 2, and reverb effect adding circuit 3, which has the configuration shown in FIG. This means that the additional processing is performed in a time-sharing manner for the two right and left systems, respectively. According to FIG. 10, the right and left input signals obtained by distributing the digital musical tone signals from the corresponding sound source channel TG(i) are firstly
The signals are input to the right and left input terminals of the delay effect adding circuit l (see Figure 11), and a delay effect is added to each in the two-manufacturer/two-output delay effect module IM, resulting in right and left delay effect component signals. Thereafter, it is applied to adders 5 and 6 via each delay effect selection switch 4 (actually serves as a multiplier for depth adjustment. The same applies to other selection switches). It is added to the direct digital musical tone signal and becomes the left and right input signals to the chorus effect adding circuit 2. These signals are added by an adder 7, and the output of this addition is input to a one-manufacturer/two-output type chorus effect module 2M to add a chorus effect, and its right and left outputs are sent to each chorus effect selection switch 8. through adders 9 and lO, respectively.
Here, it is added to the signal from the delay effect adding circuit 1, which is an upstream effect adding circuit, and becomes the left and right input signals of the reverb effect generating circuit 3 at the next stage. Furthermore, these signals are added by an adder 11, and the output of this addition is 1
The input is input to an input/two-output type reverb effect module 3M, a reverb effect is added, and its right and left outputs are given to adders l3 and l4, respectively, via each reverb effect selection switch l2, where chorus is added. It is added to the signal from the effect addition circuit 2, and output as an effect output signal from the right and left output terminals, respectively. Note that the connection order of the delay, chorus, and reverb effect adding circuits 1, 2, and 3 is not limited to that shown in the diagram;
The order can be changed by changing the program shown in FIG. 7). In that case, as can be seen from FIG. 10, the effect generating unit SE (i
) It has the advantage of changing the overall transfer characteristics, and has the characteristic that the achievable effect space becomes wider.

FIG. 11 is a block diagram illustrating the detailed functions of the delay effect adding circuit shown in FIG. 1θ. In the same figure, the delay effect module IM as a delay effect component generating means has two independent delay effect modules for adding right and left delay effects.
Shift registers la and lb forming two delay circuits and shift registers la and 1b are provided.
A clock generator (CLK) that shifts each
attenuators 1e and if that attenuate the outputs of lc and 1d and shift registers la and lb, respectively, and feed them back to the input side;
The adders Ig and lh are provided on the input sides of the shift registers la and 1b, respectively, and add the input signal and the outputs of the attenuators le and If, respectively. and shift register la. Each of the output terminals 1b has an output terminal for a delay effect. That is, the input signal is delayed by shift registers la and lb having a feedback loop, a predetermined delay effect is added, and the signal is output in stereo. Here, the shift time of the shift registers la and lb means the delay time of the delay effect, and the attenuator 1e. 1
The amount of attenuation of f means the amount of feedback of the delay effect. In addition,
As shown in FIG. 11, coefficient multipliers 4-R and 4-L are shown here instead of the selection switch 4. These coefficient multipliers 4-R and 4-L are elements that adjust the amplitude of the delay effect component signal, and the coefficients DDEPTHR. DDEP
When THL (see effect parameter memory 26 in FIG. 9) is zero, it naturally outputs zero and functions as the selection switch 4 in the OFF state. To achieve this, when there is an input from the operation panel 19 or the like in FIG. 6 to inhibit a specific effect (for example, a delay effect), the CPUIO sets a prohibition flag on the internal RAM, and
For all units of P, DDEPTHR=DDEPTH
Data with L=0 is sent.

FIG. 12 is a functional block diagram showing an example of the chorus effect adding circuit 2 shown in FIG. 1θ. In the figure, a chorus effect module 2M, which is a chorus effect component generating means, has shift registers 2a and 2b forming two delay circuits for right and left outputs having a common input, and shift registers 2a and 2b, respectively. Voltage controlled oscillators (VCO) 2c and 2d supply the modulation frequency, and one voltage controlled oscillator 2C is passed through a phase inversion circuit 2e, and a low frequency output is sent directly to the other voltage controlled oscillator 2d to determine the modulation depth. Low frequency oscillator (LFO) supplied via volume 2f
)2g. The output terminals of the shift registers 2a and 2b each have a chorus effect output terminal. That is, the low frequency output generated by the low frequency oscillator (LFO) 2g is applied to the inversion circuit 2e on one side, and directly to the shift registers z & 2b via the voltage controlled oscillators 2C and 2d on the other side.
) The oscillation frequencies of 2c and 2d are changed, a frequency modulation effect is added, and the signal is output in stereo. Of course, FIG. 12 is a functional diagram for adding the j-las effect, and the circuit elements described above (eg, voltage controlled oscillator) are realized digitally.

FIG. 13 is a functional block diagram showing an example of the reverb effect adding circuit shown in FIG. 10. In the figure, a reverb effect module 3M as a reverb effect component generating means includes one shift register 3a and this shift register 3a.
a clock generator (CLK) 3b for shifting a, and an adder 3 for adding and outputting the outputs of a plurality of intermediate taps of the shift register 3a for right and left outputs, respectively.
It is composed of C and 3d. The output sides of these adders 3C and 3d each have an output terminal for a reverb effect. That is, the input signals are variously delayed outputs from the intermediate taps of the shift register 3a and are added by adders 3C and 3d, respectively, and a predetermined reverb effect is added thereto, and output in stereo.

Next, a flowchart (see Fig. 14) explains how the various functions described in relation to Figs.
This will be explained in detail with reference to FIGS.

FIG. 14 shows the parameter reception process of the effect generating unit SEi, that is, the operation of the above-mentioned P S/S E interface 37, while FIG. 15 shows the effect adding process of the effect generating unit SEi, that is, the control circuit. 23 shows the overall digital processing performed on musical tone signals under the control of 23.

As mentioned above, when a command is sent from CPUIO (YES at TI). The decoding is performed at the interface 37 (T2). The control circuit 23 is connected to the internal bus 3
2 is not in use, the data (effect parameters) included in the command is stored in the effect parameter memory 25.
, 26 (T3).

For the CPU 110, this command transfer process is nothing but an effect parameter update process for the effect generating device SEP. In this regard, the CPU 110 in FIG. 6 performs voice assignment when a pronunciation message is given from an external performance controller, and assigns pitch information FT and touch information VL to instances of pitch information FT and touch information VL included in the pronunciation message. /or A conversion function is calculated according to the effect control function assigned to the touch information VL based on the function assignment on the operation panel 19, and the target effect parameter (DTIMER.DTIMEL.D
RPEATR. DRPEATL. DDEPTHR, DD
EPTHL, CDEPTH. CDTIME, RTIR-
DTmR. DTIL ~DTmL. RDEPTH. LF
At least one effect parameter among O parameters, etc.) is generated and the effect is assigned.
The generated data is transferred to the interface 37 of E(m). Below, the effect generating unit will be explained.
It becomes clearer from the description of the effect addition process in (i),
Effect generation 22} SE (i) digitally processes the digital musical tone signal from the corresponding sound source channel TG (i) according to the effect parameters updated at the time of performance to form a sound effect signal. As a result, sound effects can be controlled in real time according to the pitch and touch of the performance.

Effect generation unit} SEi effect addition processing flow (first
As shown in Figure 5), the sampling clock C is input externally.
When KI is applied, the F/F 24 (FIG. 7) is set, and the control circuit 23 detects that the state F becomes F=1 (step Sl). As a result, the control circuit 2
3, the program counter 228 is supplied with an eight count start signal. As a result, the program counter 22 starts counting in synchronization with the clock signal CK2, supplies the address to the program memory 2l, and the program is supplied from the program memory 21 to the control circuit 23, thereby controlling each part. become. That is, in step S2, a reset signal is supplied from the control circuit 23 to the flip-flop 24, and the flip-flop 24 is reset (CF=0). Next, in step S3, the digital musical tone signals from the corresponding tone source channel TGi stored in the input register 36 are written to WAVER and WAVEL, respectively. Next, in steps S4 to S6, delay effect addition processing (DELAY), which will be described later. Processing of adding chorus effect (CHORU
S). Processing for adding reverb effects (REVERB) is performed sequentially, and the sono result is converted to EWAVER. Finally, in step S7, EWAVER and EWVAL are transferred to the right output register 38 and left output register 39, respectively. This is shown in Figure lθ, delay effect adding circuit l, chorus effect adding circuit 2,
Add each effect with the reverb effect addition circuit 3,
Corresponds to stereo output from the output terminal.

The flowchart shown in FIG. 16 shows details of the delay effect adding process shown in step S4 in FIG. 15. In step Sll in the same figure, the incremented value of DPOINTR is ANDed with DERIAAR, and the value obtained by ORing that value with DERIAOR is set as DP.
Store in OINTR (DPOI NTR←(DPOIN
TR+1)nDERIAARuDERIAOR). Also, the contents of DPOINTR are set in the address register 34 (address register←DPOINTR). That is, in the logical operation of this step Sll, DPOINTR
When the incremented value is within the memory usage area of the delay effect, the incremented value is added to the new DPOINTR, that is, the delay memory (address DE
memory of size DERIAAR starting from RIAOR)
The pointer DPOINTR becomes the current pointer (the address that stores the current waveform data) for the current waveform data, and when it exceeds the final address of the memory, the value returned to the first address becomes the new DPOINTR (as a result, the pointer DPOINTR samples the data on the right delay memory). Performs loop access to the right delay memory, advancing by one address at each time, and returning to the start address when the end address is reached.Similar loop access is performed for other pointers to the corresponding memory. ). Next, in step S12, WAVER and DRDATA
The value obtained by adding R is set in the data register 35.

And address register 34 (DPOINTER)
Waveform data memory 33 (delay memory) instructed by
Write the value of register 35 to the address. This corresponds to adding the attenuation amount of the output of the shift register la by the attenuator 1e and the value of the input data in the adder 1g in the first 11N, and inputting the result to the shift register 1a again. Next, in step S13 of FIG.
The value obtained by adding I) TIMER to OINTR and DERIA
AND with R, OR the value with DERIAOR, and set the value in the address register 34 (address register ← (DPOINTR + DTIMER) nD
ERIAARUDERIAOR). This step Sl3
The logical operation of is similar to the processing of step Sll, but D
By jumping the address by a value corresponding to TIMEH, an address is calculated for reading past waveform data from the delay memory (delayed waveform data corresponding to the output data of the shift register 1a). In this example, I) ERIAAR-D
The value of TIMER corresponds to the original delay time. this is,
The waveform contained in the address after DTIMER is actually DE
This is understood from the fact that only RIAAR-DTIMER is a past waveform. Then, in step S14, the value of the data register 35 (delay sound effect component signal corresponding to the output of the right delay effect module IM) is set to the depth DDE.
The value obtained by multiplying the value multiplied by PTHR and adding WAVER (direct digital musical tone signal) (right delay sound effect signal) is stored in WAVEH, and the value obtained by multiplying the value of the data register 35 by the feedback attenuation rate DRPEATR is stored in DRDATAH ( WAVER +WAVER+Data register XDDEPTHR.DRDATAR←Data register x
DRPEATT). That is, the waveform data memory 3 specified by the address register 34 in step S13 above
Read the waveform data of 3 and obtain the delay sound effect for the right channel.

Next, processing similar to steps Sll to S14 and AR described above is performed for the left channel to obtain a delay sound effect for the left channel.

The flowchart shown in FIG. 17 shows details of the chorus effect adding process shown in step S5 of FIG. 15. In step S21 in the figure, processing is performed for a low frequency oscillator (LFO) to obtain waveform data for low frequency oscillation. The outline of the processing in step 321 is to store the waveform to be generated as time information, angle information, and angle change amount information, change the readout speed by the counting means and the accumulating means, and output the integer part of the waveform (LFOH). It outputs a fractional part output (LFOL) and can generate a waveform with little distortion depending on the frequency, and it is easy to obtain a fractional part output (LFOL) with a constant amount of change. be. That is, after the processing in step S21, the integer part output (LFOH) of the relative address (from the reference delay position) necessary to calculate the address of the waveform data to be accessed in the chorus memory and the waveform compensation calculation are performed. The necessary fractional part output (LFOL) can be obtained.

Next, in step S22, the incremented value of CPOINT is ANDed with CERIAA, and the ORed value of that value and CERIAO is written into CPOINT (cPoINT←(CPOINT+1) nC
ERIAAUCERIAO), and sets the contents of CPOINT in the address register 34 (address register←CPOINT). That is, when the value obtained by incrementing CPOINT is within the memory use area of the chorus effect (chorus memory), the incremented value is set as CPOINT, and when the value exceeds the final address of chorus memory, the first address is set as the contents of CPOINT. Update input pointer. Next, step S2
3, a value obtained by adding WAVER and WAVERL (corresponding to the sound effect signal from the delay effect adding circuit 1) is set in the data register 35. Then, the value of the data register 35 is written to the address of the waveform data memory 33 (chorus memory) specified by the address register 34 (CPOINT). This corresponds to the input skip processing of the chorus effect module 2M in which the outputs of adders 5 and 6 are added by adder 7 in FIG. next,
In step S24, CPOINT, LFOH and C
The value obtained by adding DTIME and CERIAA is ANDed, and the value obtained by ORing that value and CERIAO is stored in the address register 34 (address register ← (CP
OINT+LFOH+CDTIME)rlcERIAA
UCERIAO). Also, the value of the data register 35 is 1.
Multiply the value obtained by subtracting LFOL from 0, and convert the multiplied value to E
Store in WAVEH (EWAVER+data register X (1.0-LFOL).

Next, in step S25, CPOINT and LFO
The value obtained by adding H, 1, and Cl:jTIME and CERIA
Take an AND with A, and set the value obtained by ORing that value and CERIAO in the address register 34 (address register ← (CPOINT + LFOH + 1 + CDTIM
E) rl (CERIAAUCERIAO). Also, the value obtained by multiplying the value of the data register 35 by LFOL is
Add ER and store it in the value IEWAVER (EWAV
ER4-data register XLFOL+EWAVER).

In this way, step S24 and step S25cy)
For logical operations, LFOH and CDTIME are added to CPOINT.
(center delay time) and an address that adds 1 to that value. In other words, two addresses are calculated to read two adjacent waveform data on the chorus effect memory, and the two waveform data are Further, as shown in FIG. 12, the target waveform data (chorus effect component signal) is linearly interpolated using these two waveform data and the address decimal value (LFOL).

Next, in step S26, the output WAVER of the upstream effect addition circuit is added to the value obtained by multiplying the right chorus effect component signal EWAVEH by the depth CDPTH, and the added value is stored in WAVEH (right chorus effect sound component). In this manner, in steps S24 to 526, the read address is changed in accordance with the low frequency waveform, and the delay time is changed to obtain the chorus effect added sound for the right channel which outputs waveform data.

Next, in steps S27 and 328, as in steps S24 and S25 above, minus LFOH and CDTIME are set to the current C PO I NT
Calculate the address added to and the address subtracted by 1 from that value, read the data at these two adjacent addresses (on the chorus memory), and combine these two waveform data and the waveform address decimal value (LFOL). Then, the desired left chorus effect component signal is linearly corrected. That is, in steps S27 and 328, reading is performed by specifying an address corresponding to the inverted value of the output of the low frequency oscillator (LFO) for the right channel in steps S24 and S25, and then the above Compensation calculations are also performed in the same way. This means that in FIG. 12, the output of the low frequency oscillator 2g is connected to the inverter 2e.
, the other corresponds to applying the signal 5l directly to the shift registers 2a and 2b via the voltage controlled oscillators 2c and 2d, respectively, and reading it out while changing the delay time. Next, in step S29, E
WAVEL is added to the value obtained by multiplying WAVEL by CDEPTH, and the added value is stored in WAVEL. Therefore,
In steps 327 to S29, the read address is changed in accordance with the low frequency waveform of the LFO, and the delay time is changed to obtain a chorus sound effect for the left channel that outputs waveform data.

The flowchart shown in FIG. 18 shows details of the reverb effect adding process shown in step S6 of FIG. 15. In step S31 in the same figure, the value obtained by incrementing RPOINT and RERIAA are ANDed,
RPOIN is the value obtained by ORing that value with RERIAO.
Store it in T (RPOINT ←(RPOINT+l)n
RERIAALIRERIAO). or RPOINT
Store the contents of RP0INT in the address register 34 (address register ← RPOINT). In this way, when the value obtained by incrementing RP0INT is within the memory usage area of the reverb effect, the incremented value is stored in the address register 34.
When the last address of the memory is exceeded, the pointer RPOINT to the reverb memory is updated with the start address as RPOINT. Next, in step 832', "0" is stored in EWAVER and WAVER. ! =WAVEL (: 1 - right and left musical tone signals which are the processing results of the rath effect) and the added value is transferred to the data register 3. This corresponds to the process of adding the outputs of addition values 9 and 10 in the adder 11 in FIG. 10 and inputting the result to the reverb effect module 3M. and,
The value of the data register 35 is written to the address of the waveform data memory 33 (reverb memory) indicated by the address register 34 (RPOINT). Next, step S3
3, RPOINT (indicating current position) and DTI
The value obtained by adding R (delay time data corresponding to the first intermediate tap) and RERIAA is taken, and the value obtained by ORing that value and RERIAO is set to the address register 3.
4 (address register = (RPOI NT+
DTIR) 11RERIAAURERIAO)
Get the address for the intermediate tap of data register 3
Apply EWAVER to the intermediate tap waveform data imported in 5.
Store the added value in EWAVEH (EWAVER
4-EWAVER+data register). That is, in the logical operation in step S33, an address is specified for reading out the waveform data of the reverb effect memory in the area to which an address corresponding to the delay time DTIR has been added, and the waveform data memory 33 at the specified address is read out. It is stored in the data register 35 and added to EWAVEH. Next, in the same manner as step S33, the delay time DT2 is
The waveform data of the reverb effect in the area to which the address corresponding to R''D T n R has been added is sequentially read out and added. In the above, in FIG. 13, each delay output from the intermediate tap of the shift register 3a is 3
The addition at C corresponds to forming a right reverb effect component signal.

Next, in step S34, the right reverb effect component signal EWAVER multiplied by the depth RDEPTH is
Add AVER and store in EWAVEH. As a result, a reverb effect output for the right channel is obtained. Next, processing similar to steps 532 to 334 described above is performed to obtain a left channel reverb effect output for the left channel. Actually, EWAVER at this time. EWAV
The content of EL is a musical tone signal to which delay effects, chorus effects, and reverb effects are compositely multiplexed and added according to the functions shown in FIG.
This is clear from the description of the flowcharts in FIGS. 15 and 16 to 18, which are explained in association with the functional diagram in FIG. 13. These left and right composite effect signals EWAVER and EWAVE
Lt* As shown in FIG. 15, the signal is transferred to the left and right output registers 38 and 39 (FIG. 7) in step S7, and is output to the outside.

From the above description, the structure, operation, and effects of the second embodiment are clear. For example, for pitch information and/or performance touch, the delay effect feedback amount parameter DRPEA
T.R. If the function to control DRPEATL is assigned, when the CPUIIO receives the sound generation message from the performance controller, it will set the parameters corresponding to 4-5 and 4-6 in Figure 4 according to the contents of the function assignment flag. An update process is performed, and a feedback amount parameter is generated from the received pitch information Pi and/or performance touch VL, and is transferred to the effect generating unit }SE(m) of the effect generating device SEP to which the voice is assigned. As a result, address 14 (DREATR) and address 15 (DR
EATL), generated in CPUIIO,
The feedback amount parameter transferred from is set (
(See Figures 7 and 14). After the sound generation instruction, a digital musical tone signal as a direct sound is output at each sampling time from channel TG(m) of the voice-assigned sound source 140, and the corresponding effect generating unit }SE(m)
) is set in the input register 36. Then, the effect generation unit SE(m) is
Following the flow shown in Figure 5 (and Figures 16 to 19),
The digital musical tone signal transferred to the input register 36 is processed to generate left and right digital sound effect signals, and set in the left and right output register names 38 and 39. These signals are then provided to the left and right channels of a stereo sound system (not shown) through left and right adders 111, 112 in FIG. Here, the effect generating unit}S
E(m) is the delay effect addition processing 54 (Fig. 15)
During execution, as shown in step SL4 in FIG. 16, the feedback amounts DRPEATR and DRPEATL are multiplied by the past digital musical tone signals retrieved from the delay memory to obtain the net delayed feedback signals DRDATAR and DRDATAL. This is input into the delay memory in addition to the direct digital tone signal WAVEH at step S12 at the next sampling time. The above feedback amount DRPEATR,
DRPEATL has a size calculated by the CPUIIO from the pitch information FT and touch information VT specified by the pronunciation message. Therefore, a delay effect is applied depending on the value of the pitch information and/or touch information at that time, making it possible to express a performance with a wide variety of variations. If the control function assigned to the pitch information or touch information is another effect parameter, the effect is added according to the calculated value of the effect parameter. This is clear. Therefore, further explanation will be omitted.

This concludes the explanation of the second embodiment. In the case of the second embodiment, the effect generating device SEP is configured polyphonically corresponding to each channel of the polyphonic sound source 140, and therefore the sound generation message to which the CPU 110 has voice assigned Another advantage is that desired effect parameters in the effect generation device SEP (internal data in the effect parameter memories 25 and 26 of the individual effect generation units SEi) can be updated dynamically.

However, the present invention is not limited to the above-described embodiments, and includes the [claims] and [structure, operation, and development of the invention].
Various modifications and changes are possible as shown in . Furthermore, the configurations of the effect generating devices SE and SEP are not limited to those shown in the drawings, and any other appropriate musical sound signal effect generating device such as a DSP can be used. For example, in the second embodiment, all effects of delay, chorus, and reverb are applied to the seventh
Although this is achieved with one piece of hardware as shown in the figure, it is possible to use one piece of hardware for each effect or one piece of digital processing section, and connect the signals of these pieces of hardware in cascade. , a pipeline structure may be used. Conversely, instead of performing digital processing to add effects to one voice (a unit of output from one sound source channel) using one piece of hardware, time-division multiplexed polyphonic voices from a time-division polyphonic sound source can be processed using a common hardware. (If the amount of data processing is relatively small, DSP implemented on today's LSI or VLSI chips may be used.)
).

In addition, the adder itt, 112 on the output side of the effect generation device SEP of the second embodiment adds the left outputs of all the effect generation units }SEO-SEn and the right output, considering two-channel stereo. is added, but it is not necessarily necessary. Alternatively, an adder that performs addition for each musical tone system (timbre) may be used, and a panning effect generating section may be provided in the sound system, so that the panning effect can be added systematically there. In addition, the effect generator SE
P may be one that adds different effects to different musical tones. From the CPUIIO point of view, this means that
The internal parameters of each effect generating unit (SEO to SEm) are parameters that depend on the tone used by each unit, which is which of the multiple tones selected for the entire system, and parameters that depend on pitch information/touch information. , or parameters that depend on both.

[Effects of the Invention] As is clear from the above detailed description, in claim 1, at least one effect parameter used by the digital sound effect generation means is generated according to a characteristic performance parameter of a performance operation, and the sound effect is generated. Since it is supplied to the generating means, it is possible to dynamically control the characteristics of the musical sound effect according to the performance, expanding the possibilities of performance expression by the performer, and allowing for a very high degree of freedom of change never seen before. Making it possible to express a rich musical space.

Further, in claim 2, at least one effect parameter used by the sound effect component generating means is controlled according to a performance parameter, and the sound effect component signal whose characteristics are controlled and output from the sound effect component generating means is used as a musical sound as a direct sound. Since it is output together with the signal, the component of the sound effect added to the direct sound can be changed according to the intention of the performance at that time.

Furthermore, in claim 3, since the playing touch and/or pitch are employed as the performance parameters, it is possible to control how the sound effect is applied based on touch and pitch, which are the basics of musical instrument performance.

Further, in claim 4, since the musical sound signal as the direct sound and the sound effect component signal are added and output, there is an advantage that the number of output lines is reduced, and when outputting to the outside through the sound system, A combination of direct sound and sound effect components can be emitted from one speaker.

Further, in claim 5, the coefficients for the counting multiplication means, which is a component of the sound effect generation means, are set according to the performance parameters, so that the depth of the effect, the amount of feedback of the internal signal, the periodicity of the effect, etc. are transmitted. Characteristic elements can be freely controlled according to the performance.

Further, in claim 6, the sound effect generating means is a plurality of effect generating module means (sound effect component modules) which are continued in cascade.
The output of the upstream module means is adjusted in depth between adjacent module means, and the input signal of the downstream module means is obtained by adding the depth-adjusted output of the upstream module means and the input of the upstream module means. Since a means for forming the effect is provided and the depth to be adjusted is controlled by the performance parameter, the depth at which the effect is applied can be changed for each module.

Further, in claim 7, the relative magnitude between the effect sound component signal which is the output of the effect component means and the musical tone signal as the direct sound is controlled according to the performance parameter.
You can change the volume balance between direct sound and sound effects to suit your performance. In this case, the final circuit element for changing the amplitude of the signal can be realized not only by digital circuits but also by analog circuits.

Further, in claim 8, a coefficient multiplication means for multiplying the musical sound signal by a coefficient is provided on the input side of the effect component generation means, and this coefficient is controlled by performance parameters, so that the effect component generation means can generate a signal from a polyphonic sound source. Even when a plurality of musical tone signals are processed as one total musical tone signal, the magnitude of each sound effect component signal can be controlled according to individual performance parameters.

Further, in a ninth aspect of the present invention, the coefficients of the coefficient multiplication means provided on the output side of the effect component generation means are controlled by performance parameters. In this case, if the effect component generating means is configured to correspond to the channels of the polyphonic sound source, the magnitude of each effect sound component signal can be controlled according to each performance parameter. Furthermore, when the effect component generating means is of a type that performs digital calculations using a fixed point, there is an advantage that the numerical accuracy is higher than when the signal is scaled on the input side.

[Brief explanation of the drawing]

FIG. 1 is an overall configuration diagram of an electronic musical instrument according to a first embodiment incorporating the features of the present invention, FIG. 2 is a configuration diagram of the mixer 15 shown in FIG. 1, and FIG.
The configuration diagram of the mixer l7 shown in the figure, Fig. 4 is the performance touch VL,
A flowchart of the processing operation of the CPUll for controlling the relative amplitude ratio VEL of the reverberant sound according to the pitch FT, FIG. 5 is a diagram illustrating a conversion function when converting the playing touch and pitch into the relative amplitude ratio VEL, and FIG. 6 7 is a block diagram of the hardware of the effect generation unit SEi shown in FIG. 6, and FIG. 8 is a block diagram of the effect parameter memory 25 of FIG. 7. 9 is a diagram showing the internal configuration of the effect parameter memory 26 in FIG. 7, FIG. 1θ is a functional block diagram of the effect generation unit in FIG. 7, and FIG. 11 is a delay diagram in FIG. FIG. 12 is a functional block diagram showing an example of the chorus effect adding circuit in FIG. 10; FIG. 13 is a functional block diagram showing an example of the reverb effect adding circuit in FIG. 10. , FIG. 14 is a flowchart showing the operation of the CPU/SE interface in FIG. 7, FIG. 15 is a flowchart showing the effect addition processing operation performed under the control of the control circuit in FIG. 7, and FIG. 1
5. FIG. 17 is a flowchart showing details of the chorus effect addition process in FIG. 15. FIG. 18 is a flowchart showing details of the reverb effect addition process in FIG. 15. FIG. 19 is a diagram illustrating the calculation process of the chorus effect. l l...CPU, 12...I'
lOM, 1 3...RAM, SE...
- Reverberation effect generator, l5...first mixer, 16-...reverberant sound generator, 17...second mixer, FT...performance Pitch, VL...
...Performance touch, 1.10...CPU, S
EP・・・Effect generating device, SEO'-SE n・
...Effect generating unit, 25, 26...
Effect parameter memory, IM...Delay effect generation module, 2M...Chorus effect generation module, 3M...Reverb effect generation module. Patent applicant Casio Computer Co., Ltd. VEL(n) Fig. RVL Fig. Fig. 4 Sitter face 37 operations Fig. 14 SEi6)'? Shiokai size 1 bite Kansyu F 15th cause

Claims (9)

[Claims] (1) A sound effect generating means that receives a musical sound signal generated based on a performance operation in the form of digital data, and generates a sound effect signal by digitally processing the input musical sound signal according to internal effect parameters. , an effect parameter control means for generating at least one of the effect parameters used by the sound effect generation means in accordance with a performance parameter characterizing the performance operation and supplying the effect parameter to the sound effect generation means. Sound effect control device. (2) A musical sound signal generated based on a performance operation is input in the form of digital data, and an effect component generating means generates a sound effect component signal by digitally processing the input musical sound signal according to internal effect parameters. and output means for outputting the sound effect component signal together with the musical sound signal component; generating at least one of the effect parameters used by the effect component generation means in accordance with a performance parameter characterizing the performance operation; A sound effect control device comprising: an effect parameter control means for supplying the effect parameter to the means; (3) The sound effect control device according to claim 1 or 2, wherein the performance parameter is at least one of performance touch and pitch. (4) The sound effect control device according to claim 2, wherein the output means includes addition means for adding the musical tone signal and the sound effect component signal. (5) In the sound effect control device according to claim 1, the sound effect generation means includes a plurality of coefficient multiplication means for multiplying the signal being processed by the effect parameter as a coefficient, and the effect parameter control means A sound effect control device characterized in that a coefficient used in at least one of the coefficient multiplication means is generated according to the performance parameter. (6) In the sound effect control device according to claim 1, the sound effect generation means includes a plurality of cascade-connected effect generation module means capable of adding different effects in a composite manner, and the adjacent effect generation module means. A depth adjustment multiplication means provided between and multiplying the module output signal of the upstream effect generation module means by a depth parameter as the effect parameter, and a module whose depth is adjusted by the depth adjustment multiplication means. addition means for adding the output signal and the input signal of the effect generation module means on the upstream side to form an input signal for the effect generation module means on the downstream side; A sound effect control device comprising: depth parameter generation means for generating the depth parameter used by the multiplication means in accordance with the performance parameter. (7) an effect component generating means for generating a sound effect component signal by receiving a musical sound signal generated based on a performance operation in the form of digital data and digitally processing the input musical sound signal; output means for outputting the signal and the sound effect component signal; and effect depth control means for controlling the relative magnitude of the sound effect component signal with respect to the musical tone signal according to a performance parameter characterizing the performance operation. A sound effect control device comprising: (8) In the sound effect control device according to claim 7, the effect depth control means includes: a coefficient multiplication means for multiplying the musical sound signal input to the effect component generation means by a coefficient; A sound effect control device comprising: coefficient generating means for generating coefficients in accordance with the performance parameters. (9) In the sound effect control device according to claim 7, the effect depth control means comprises: a coefficient multiplier for multiplying the sound effect component signal outputted from the effect component generation means by a coefficient; A sound effect control device comprising: coefficient generating means for generating the coefficient according to the performance parameter.