JPH03225398A - Control method for sound source for electronic musical instrument - Google Patents

Abstract

PURPOSE:To enable performance in an invariably normal state by correcting the operation data of a performance operation element corresponding to musical sound parameters according to sound generation range characteristics corresponding to the musical instrument and then inputting the corrected data to a sound source circuit. CONSTITUTION:The position information based upon the operation of the performance operation element 1 is converted through an A/D converter 2 and an arithmetic circuit 3 for speed conversion into speed data iv, which is inputted to a correcting circuit 4. Further, pressure information from the pressure sensing means of the operation element 1 is inputted as pressure data ip to the circuit 4. The circuit 4 corrects the data iv and ip at the start of 1st sound generation so that the data enter a sound generation area where a sound begins to be generated, and then inputs the corrected data as bow speed data vv and bow pressure data vp on a violin, etc., to the sound source 6. Further, the pitch information (p) corresponding to a scale is inputted to the sound source 6 through operation on a keyboard 5 and the sound source 6 generates an electronic sound according to those respective data, so that the sound is outputted through a sound system 7.

Description

[Detailed Description of the Invention] [Industrial Application Field] The present invention relates to a method of controlling a sound source for an electronic musical instrument, and in particular, the present invention relates to a method of controlling a sound source for an electronic musical instrument, and in particular, the sound source circuit is controlled based on input data such as position and pressure corresponding to musical tone parameters from a performance operator. This is an improvement to ensure that the words are always pronounced correctly.

[Prior Art] Electronic musical instruments that generate the performance sounds of bowed string instruments such as violins and wind instruments such as clarinets use mechanical vibrations of the strings corresponding to the movement of the contact point between the strings and the bow, and air vibrations within the mouthpiece of the wind instrument. It is equipped with a physical sound source that generates an electronic sound that physically approximates the sound produced by an electric circuit using an electric circuit. In such electronic musical instruments, in addition to inputting pitch information of pressed keys by operating the keyboard, performance controls such as slide volumes are used to input parameters corresponding to bow pressure and bow speed for string operation, design of blow operation, and amplifier sure. Control signals are input to the sound source to create and generate electronic sounds.

In conventional electronic musical instruments, a musical tone control signal based on the operating position and operating pressure of a performance operator is multiplied by a certain coefficient regardless of the speed, pressure, etc., and is essentially input directly to the sound source.

[Problems to be Solved by the Invention] However, when the operation information of the performance controllers is directly input to the sound source, there are cases where no sound is produced in a certain operation area, or an irregular sound such as an unpleasant sound or a so-called inside-out sound is generated. There was a flash. Therefore, it is necessary to operate the performance controls while avoiding the occurrence of such irregular sounds, making it difficult to play the electronic musical instrument.

The reason for such irregular sounds is that, in the case of bowed stringed instruments, for example, the parameter relationship between bow pressure and bow speed does not fall within the normal sound production range. The relationship between bow pressure and bow speed for a stringed instrument is as shown in the '1fi2 diagram. Four straight lines passing through the origin define the normal sound generation area A where the sound begins, and the sustaining area B where the sound generated at -degrees continues. and an irregular sound region C where the sound disappears or an unpleasant sound is generated.

Therefore, when the performance operator is operated in a state corresponding to a certain bow speed (■1), if the bow pressure at that time is too high or too low and does not fall into the sound generation region A, the sound will not begin to sound. Also,
If the sound enters the irregular sound region C, the sound disappears, or an unpleasant sound or a reversing sound is generated.

In conventional electronic musical instruments, the operation information of the performance controls is essentially input directly to the sound source, so depending on the operating conditions, the sound may fall into an irregular sound range.In such cases, the sound may disappear or an unpleasant sound may be produced. It was occurring.

The present invention has been made in view of the above-mentioned drawbacks of the prior art, and an object of the present invention is to provide a sound source control method for an electronic musical instrument that can always be played in a normal sounding state regardless of the operating state of the performance operator.

[Means for Solving the Problems] In order to achieve the above object, the present invention corrects operation data of performance operators corresponding to musical tone parameters based on sound generation area characteristics corresponding to the musical instrument, and then uses the corrected data. is input to the sound source circuit.

These sound generation area characteristics are represented by a graph consisting of four curves in a coordinate system with the two musical tone parameters as the vertical and horizontal axes.Substantially, the area between the two central curves constitutes the sound generation area. The regions on both sides constitute a sound sustaining region, and the region outside the two outermost curves constitutes an arrhythmia region.

[Operation] When the operation data from the performance operator is in the irregular tone area, for example, it is corrected to data that falls within the sound generation area, and this corrected data is manually input to the sound source.

[Examples] Hereinafter, the present invention will be described in more detail based on Examples with reference to the drawings.

FIG. 1 is a block diagram of an electronic bowed string instrument according to the present invention. The performance operator 1 includes, for example, a slide volume equipped with pressure-sensitive means, a joystick mechanism, or a mouse mechanism. Position information obtained by operating the operator 1 is converted into speed data (iv) via an A/D converter 2 and a speed conversion arithmetic circuit 3, and then manually input to a correction circuit 4. Further, pressure information from the pressure sensing means of the operator 1 is inputted to the correction circuit 4 via the A/D converter 3 as pressure data (ip).

At the beginning of the first sound, the correction circuit 4
The velocity data (iv) and the pressure data (ip) are corrected so that they fall within the sound generation region where the sound begins, and are input to the sound source 6 as bow velocity data (vv) and bow pressure data (vp). Pitch information (p) corresponding to a musical scale is further input to the sound source 6 by operating the keyboard 5. The sound source 6 creates an electronic sound based on these bow speed data, bow pressure data, and pitch data, and emits the sound via the sound system 7.

The calculation method for correcting the sound production area in the correction circuit 4 will be explained below.

As described above, the relationship between the bow speed VV and the bow pressure Vp, which are parameters for controlling the musical sound of a bowed string instrument, is shown in FIG. 2. Four straight lines a, b, c, and d pass through the origin, and their slopes differ depending on the pitch information from the keyboard. Therefore, the correction calculation program is performed based on a table using key numbers. Each straight line also differs depending on the distance from the top of the stringed instrument to the string position. Therefore, the calculation program includes a table that uses the distance from the top as a parameter. Also, 3rd gear has positive and negative values, but since the graph is symmetrical about the vertical axis (3rd pressure), it will be omitted.

3 pressure VP to ensure proper sound regardless of 3rd gear
An example of a calculation formula for correcting and positioning within area A is shown below.

vv=iv...Glance...■vp=((c+b)/2)xiv...■By such calculation, if vv is not zero, it will definitely fall within the sound generation area A and the sound will begin to sound. In this case, the volume changes depending on vv, but the sound quality is constant and monotonous. In addition, pressure data (i
p) is ignored. Note that c and b represent the slopes of the straight lines c and b.

Next, a calculation formula for changing the sound quality using pressure data (ip) will be shown.

vp-bx iv+, (c-b) x ivx ip/
pma m+ + + ■The above ■ formula changes the sound quality by changing the position in the vertical direction within the sound generation area A based on the pressure data from the controller *pHaX is the maximum value of the input pressure data (ip) be.

After the physical sound source that generates the electronic sound generates the -dan sound,
It has a hysteresis characteristic that maintains normal pronunciation of sounds even in the sustaining region B outside the sounding region A. Therefore, after the -dan sound begins to sound, it is desirable to vary the position of the parameter widely, including the sustain region, to provide a range of sound quality.

A calculation formula for expanding the range of such sound production area is shown below.

vp=ax iv+(d-a) x ivx ip/P
, , , ... By controlling the sound source using the 00 formula and the ■ formula, it is possible to produce normal sound over a wide range and to increase the change in sound quality and volume.

FIG. 3 shows a flowchart for switching between sustain processing in which the control region is expanded to sustain region B and rise processing for starting the sound.

The number of times the start-up process is repeated after start-up is set in advance. In step 301, it is determined whether the counter has reached this set value. If it is less than the set value, the rising process is repeated (step 302). After processing, increment the counter and increase the number by 1 Step 3
04), return to step 301. When the counter reaches the set value, the sustaining process is performed (step 303), and when the -1st sustaining process is performed, the counter is incremented (step 30).
4), the persistence process is repeated via the determination step 301. The counter is cleared to 0 when a new sound is generated, for example, when a key-on signal is input or when the pressure data (ip) of the operator changes from 0 to 1.

In the above correction calculation, ip% iv is directly set from 0 to
Since it is passed through a table based on key numbers etc. without being used for inputting the 0 type, it is possible to control musical tone parameters in line with human senses.

Further, in the above correction calculation, ip is created from iv, but iv can also be created from ip. As examples of such calculations, the following equations (2) and 0 are shown in correspondence with the above equations (2) and (2), respectively.

vp=vi ・ ・ ・・■ vv= (1/d) X ip” ((1/a) −(
1/d) )x ipx iv/V□8...■ Here, V@H1 is the maximum value of iv. You can select each calculation formula according to your playing feeling depending on the instrument.

Each straight line a, b of the conversion characteristic graph shown in FIG.

By adjusting the slopes a, b, C, and d of c and d, it is possible to widen the range of emotional expression by adjusting the range of each area and slightly including the irregular blue area C. in this case,
The conversion characteristics may be settable during performance.

FIG. 4 is a straight line a showing an example of a characteristic graph for pronunciation correction calculation of a bowed string algorithm for an electronic bowed stringed instrument using a three-dimensional tablet as an input device.

The area between 0 and 0 is the sound generation area A where the sound begins, the area between the straight lines C and d is the sustaining area B where the sound continues, and the area outside the straight lines C and d is the irregular blue area C. The slope of each straight line is a, b, c,
d changes depending on the stringing point (distance 1 m1 from the top). It also changes depending on the key number. In particular, d varies greatly depending on the key number. In such a characteristic graph, the sound will not start unless it enters sound generation area A at the time of start-up.The calculation formula for correcting 3rd speed from 3rd pressure to ensure that it enters sound generation area A at start-up is shown below. .

vb=fb/c+((1/a)-(1/c)) ×v
b/V+ma Xxfb where vb and fb indicate 3rd speed and bow pressure, respectively.

After the sound starts playing, enter the sustain area B. The range below the straight line d is where the sound disappears, and the range above the straight line d is where the unpleasant sound is generated. The calculation formula for correcting the 3rd speed based on the bow pressure and ensuring that it enters the sustaining region B after starting up is shown below.

vb-fb/d◆((1/b)-(1/d)) xvb
/v□, xfb FIG. 5 is a block diagram of a control mechanism of an electronic musical instrument equipped with the above correction circuit.

Signals from the performance operator 15 and the keyboard 13 are input to the CPU 18 from a pass line via a detection circuit 16 and a keyboard switch circuit 14, respectively. CPLT1B includes a program ROM 19 that stores each routine program, a data ROM 20 that stores data necessary for arithmetic processing, and a work RAM 21 that stores the results of each calculation during arithmetic processing.
The necessary data is read out from the above and the musical tone control parameters subjected to the above-mentioned correction calculation are calculated. The function operator 22 is normally used to select timbre, vibrato, etc., and to switch various modes. For example, the function operator 22 is used to switch between bow position detection mode and 3-speed detection mode. The timer 17 interrupts the main routine of the program by the CPU 18 at a fixed cycle of about several milliseconds.

FIG. 6 shows the basic main routine. In step 8, each arithmetic circuit is initialized, and each sound source parameter is set to a predetermined initial value. Next, the keyboard key switch processing (step 9) and other switch processing (step 10)
is repeated. For such a main routine, an interrupt routine (described later) is generated at a constant cycle by the timer 17.
is executed to perform the correction calculation.

FIG. 7 shows the mode switching routine. At step 11, mode switching such as the detection mode is performed, and the detection results and the like are stored in a register for the next detection calculation process (step 12).

An example of a human-powered device for controlling musical tone parameters of an electronic musical instrument according to the present invention is shown in FIGS. 8 and 9.

FIG. 8 is a top view, and FIGS. 9(a) and 9(b) are an enlarged side view and a top view of essential parts, respectively. This input device is a slide volume type operator, and an operator 27 slides along a guide groove 26 in the center of a main body 25 that constitutes a first slide volume. As shown in FIG. 9, the operator 27 includes a slider 28 that slides along the guide groove 26 in the direction of an arrow, and an operating piece 29 attached to the slider 28. The operating piece 29 is desirably rotatable in the direction of arrow F relative to the slider 28 in order to facilitate sliding operation. In this case, the rotation angle can be detected and used as musical tone control data. Operation piece 2
9 constitutes a second slide volume. Operation piece 29
The slider 31 slides along the guide groove 30 in the direction of arrow E. First position data is obtained from a resistance value that corresponds to the position of the operator 27, and second position data is obtained from a resistance value that corresponds to the position of the operator 31.

A pressure sensor 32 is attached to the side surface of the operator 29 to measure the pressure during operation and obtain pressure data. Musical tone control parameters are calculated based on these two position data and pressure data, and the above-mentioned correction calculation is performed.

Figure 10 shows the key-on routine by the CPU.
First, the key code of the pressed key is stored in the key code register (KCD) (step 33). Next, the sound source channel is assigned. The assigned channel is stored in an assignment channel register (AC) (step 34). Next, predetermined read management data (T
A filter coefficient of a musical tone control filter circuit, which will be described later, corresponding to the key code of the KCD is read from the CD) and sent to the assigned channel (ACH) of the sound source (step 35).
). Next, the assigned channel is instructed to produce sound and is registered (step 36). At this time, a signal "1" is input to the flag of the registered channel.

FIG. 11 shows the key-off routine. First, the key code of the released key is stored in the KCD (step 3).
7). Next, the channel table is used to search for the sound source channel to which the key code is assigned (step 38). In determination step 39, it is determined whether such a channel exists or not. If not, the routine is terminated, and if so, it is determined whether all other channels are “O” (Sten 2.40). If all the flags of other channels are “O”, the routine is terminated and the flag is “0”.
If not, input “O” to the flag of the assigned channel (step 41), and then input 11& from the readout table.
! The release attenuation coefficient of the sound corresponding to the key code is read out (step 42). This attenuation coefficient is sent to the assigned channel (step 43). Next, the processing count of the assigned channel is cleared to "0" and the routine ends (step 44).

FIG. 12 shows the timbre selection routine. A timbre number transmitted from a timbre selection operator (for example, a selection switch on the instrument body or the aforementioned performance operator) is input into a predetermined register (step 45).

FIG. 13 shows an example of a register table for managing sound generation channels of a sound source. In this example, the number of channels of the sound source is four, matching the number of strings of the violin. By providing a plurality of sound sources in this way, when a key-on signal is transferred from one channel to another, the reverberation effect of the original sound source can be obtained. , when checking the channel flag in each routine, set the number i and increase it by 1 from the initial number 0 until it reaches 3.
Repeat times.

FIG. 14 shows an interrupt routine that interrupts the main routine at regular intervals using a fixed clock. First, let's start with the aforementioned slide volume type performance controls (Figures 8 and 9).
), the first and second position data and pressure data are stored in predetermined registers PO3I, POS2. It is stored in PRES (step 46), and then mode determination is performed in step 701. If the mode is "1", a conversion table (PVTBL) created and stored in advance is used to convert the bow speed V directly from the first position data (posB) of the operator.
is obtained (step 47).

If the mode is “0” in step 701, the speed is calculated from the difference between the previous and current first position data and the register (V
EL) (step 48). At this time, since the detection timing is constant, the difference in position directly corresponds to the speed. Next, this velocity data VEL is converted into bow velocity V using another conversion table (VVTBL) created and stored in advance (step 49). Once the bow speed V is determined, the current first position data PO31 is stored in a predetermined register for the next calculation (step 50).

The bow speed V obtained as described above is compared with a predetermined threshold (step 51). If it is smaller than the threshold, ignore it as noise and enter "0" for the number of processing times for each channel (
Step 52). If the bow speed V is greater than the threshold, a sound source control routine is executed using this bow speed V (step 53), and each parameter of the sound source is calculated based on human data as described later and sent to the sound source. .

FIG. 15 shows a sound source control routine. First, the routine is executed for channel "0" in the channel table (FIG. 13) described above (step 54). It checks whether the flag of this channel is "1", that is, whether it is the channel to be controlled. (Step 55). If it is not “1”, the determination is repeated for the other three channels (step 6
6).

If the flag is 1'', the key code of this channel is stored in the KCD register (step 56). Next, the attenuation coefficient data and delay length data corresponding to the key code are stored from the tone data group of the tone number (see Figure 12). (step 57). Next, the second position data PO32
Performs a predetermined operation using and stores it in register PO (
Step 58). Next, it is determined whether the number of rise processing times has reached a predetermined set value TMAX (step 59).
). If not, the sound quality area A of the timbre (pronunciation) characteristics
(See Figure 42) In order to input the control parameters into the timbre data group, the slope of the straight line corresponding to the key code, c-
b is read out and a predetermined calculation is performed to obtain C1 and C2 shown in the figure (step 61). When this calculation is completed, the number of times of processing i is incremented (step 62).

On the other hand, if the number of times of processing reaches the predetermined setting value in the determination step 59, both duration areas B of the pronunciation characteristics (see figure N2)
In order to put the control parameters within the range of , the slopes a and d-a of the straight lines corresponding to the key codes are read out from the timbre data group, and predetermined calculations are performed to obtain C1 and C2 shown in the figure (step 60 ),.

Next, three pressures P are calculated based on the above C1 and C2 (step 63), and data D1 and D2 for controlling the two delay circuits of the sound source circuit, which will be described later, are calculated (step 64). D2 is DC (two damping coefficients)
, P (3 pressure) and %V (3 speed) are sent to the processing channel of the sound source (step 65). In the figure, CD1.
CD2 indicates the delay reference length of each of the two delay circuits, and ΔDD indicates the range of change in bow position. The above process is repeated for the four channels (step 66).

FIG. 16 shows an example of a register table that stores the aforementioned timbre data group. The 6th frame indicates a read management data area, and the number of areas corresponds to the number of key codes. As shown in the figure, each management data serial includes a damping coefficient data register 68, a release damping coefficient data register 69,
Filter coefficient data register 70, delay length data register 71, data register 72 for v-P conversion at rise, and data register 73 for v-P conversion at sustain
is stored.

FIG. 17 shows an example of a physical sound source circuit for generating electronic sounds for a bowed stringed instrument. 702 and F03 indicate an adder, which corresponds to a chord point. 704 and 705 indicate multipliers, which correspond to the string ends on both sides of the string point. A closed loop consisting of an adder 702, a delay circuit 706, a low-pass filter 707, an attenuator 708, and a multiplier 704 corresponds to a string on one side of the string point, and the delay time of the closed loop corresponds to the resonant frequency of that string. Similarly,
Similarly, the closed loop consisting of adder 703, delay circuit 709, low pass filter 710, attenuator 711 and multiplier 705 corresponds to the string on the other side of the chord point. 7
Reference numeral 12 denotes a nonlinear function generator. This nonlinear function generator 12 adds a signal corresponding to the bow speed to a signal obtained by combining the outputs of closed loops on both sides of the stringing point in an adder 713, and further adds a signal corresponding to the bow speed. A signal obtained by adding a signal obtained by multiplying the signal from the fixed hysteresis low-pass filter 714 by a multiplier by a gain G is input. Further, hysteresis control of the nonlinear function generator 712 is performed using signals corresponding to the three pressures.

Next, an embodiment in which the present invention is applied to control of a sound source for an electronic wind instrument will be described. The wind instrument algorithm uses
The pronunciation characteristics based on the relationship between ampsure and design can be approximated as shown in FIG. Unlike the case of a bowed string instrument, the four straight lines do not pass through the origin and each have an intercept. As in the case of bowed string instruments, the range between the two central straight lines a and c is the sound generation area A, the range on both sides is the sustain area, and the area outside of the two outer lines a and d is the sound generation area A. The range is the arrhythmia region C.

Furthermore, the slope of each straight line differs greatly depending on the pitch.

FIG. 19 shows an example of a control system for correcting the control signal from the performance operator and inputting it to the sound source based on the sound generation area characteristics unique to wind instruments. As an input device, a performance operator 74 consisting of a three-dimensional tablet that draws a trajectory on the XY plane is used. The pen pressure data for pressing the performance operator and the XY coordinate data indicating the X position and Y position are input to the sound production area correction conversion program 76 of the detection circuit 75. The input data is converted by the conversion program flow into design data and amplifier sure data whose values fall within a predetermined normal sound generation range, and are input to the sound source control circuit 77.

On the other hand, by operating the key code number 8, the key number is input to the sound source control circuit f7 as pitch information.

As in the case of bowed string instruments, the physical sound source that generates electronic sound is
- After the dan sound is generated, the duration area B outside the sound production area A
It also has a hysteresis characteristic that maintains normal pronunciation of sounds. Therefore, after the -dan sound begins to sound, it is desirable to vary the position of the parameter widely, including the sustain region, to give the width of the front chamber.

FIG. 20 shows a flow for determining switching between the start-up process for starting the sound and the sustain process after the start-up. First, the maximum value of amplifier sure is set (step 79
). Next, four straight lines a.

The slopes and intercepts of b, c, and d (Fig. 18) are read out from a predetermined table (step 80). Next, additional intercepts for the rise and duration are calculated (step 81).

Next, in step 82, it is determined whether or not it is start-up processing. This determination, for example, counts the number of start-up processes,
This is done depending on whether the count has reached a predetermined set value. If it is a rising process, the parameters are determined by the key code (step 83), the minimum value of amplifier sure is set (step 84), the design at the rising time is calculated (step 85), and the design is corrected. (Step 86). When this design correction is completed, the count is incremented (step 87), and when the count exceeds a predetermined set value, the process is switched to continuous processing, the parameters are determined by the key code (step 88), and the minimum value of amplifier sure is determined. set (step 89)
, a design for sustaining the sound is calculated (step 90).
, the design is corrected (step 91).

FIG. 21 is a block diagram of a sound source control system for a wind instrument type electronic instrument according to the present invention. X position data, Y position data, and pen pressure data are transmitted from a performance operator 74 consisting of a three-dimensional tablet and sent to a register 94.
is memorized. X.

For the Y coordinate data, speed, direction, and distance are calculated at fixed time intervals by an arithmetic circuit 93 using the reference coordinates X o and Y o stored in the reference point coordinate register 92 , and the calculation results are stored in the register 94 . . The register 94 is connected to a tone control parameter calculation circuit 95.

On the other hand, key code information representing a musical scale and lateral vibration information for shifting each parameter value in positive and negative directions are transmitted from the keyboard 78 and stored in the register 96. This register 96 is also connected to the tone control parameter calculation circuit 95.

The musical tone control parameter calculation circuit 95 reads data necessary for calculation from the registers 94 and 96, calculates each parameter, that is, design, amplifier sure, delay length, multiplier coefficient, filter coefficient, and other data, and sends it to the sound source 97. . A sound source 97 emits the created electronic sound from a sound system 99 via a D/A converter 98.

FIG. 22 is a block diagram of the control mechanism of the electronic wind instrument according to the present invention. Signals from the performance operator 74 and the keyboard 78 are input to the CPU 18 via a pass line. C
The PU 18 reads necessary data from a ROM 103 that stores data necessary for each routine program and arithmetic processing, and a RAM 104 that stores calculation results during the arithmetic processing, and calculates tone control parameters. A panel switch 105 is used to select tone, vibrato, etc., and to switch various modes. The display 106 displays the selected switch or mode. The timer 17 is used to interrupt the main routine of the CPU 1B at a fixed cycle of about several milliseconds.

FIG. 23 shows the basic main routine. Step 10
At step 7, each arithmetic circuit is initialized, and each sound source parameter is set to a predetermined initial value. Subsequently, the keyboard key switch process (step 108) and other switch processes (step 1o9) are repeated. For such a main routine, an interrupt routine (described later) is executed at regular intervals by the timer, and the various control parameters are calculated.

FIG. 24 shows a key-on event routine when the keyboard is pressed in step 108 of the main routine. The key code of the pressed key is stored in register KCD (step 110).

FIG. 25 shows a routine when numerical input to the design control device for setting design-related parameters in step 109 of the main routine is turned on.

First, a manual numerical value is stored in register BUF (step 111). BUF data is breath pressure device register PD
The information is stored in the EV (step 112). Next, the design control device name is displayed (step 113). In this embodiment, a device (operation data) for controlling a design etc. can be arbitrarily selected.

FIG. 26 shows the routine for the sway effect switch-on event within step 109 of the main routine. In step 114, a flag indicating whether or not a lateral shaking effect is to be applied to a predetermined parameter is switched. Step 11
In step 5, it is determined whether or not this flag is at "1", that is, whether or not the lateral vibration is effective. If it is "1°," a display indicating that the lateral sway is on is performed (step 117), and if it is not "1°, a display indicating that the lateral sway is off is displayed (step 116). In this embodiment, it is possible to select for each parameter whether or not to take into account the parameters caused by the lateral vibration of the keyboard.

FIG. 27 shows the panel switch processing routine in step 109 of the main routine. When the switch is turned on, the edit screen displaying the parameters to be processed is selected and the screen number is stored in the register PAGE (step 118). This memorized number screen will be displayed (
Step 119). Next, it is determined in order whether design, amplifier sure, delay, and other edit processing is to be performed (determination steps 120, 122, and 12).
4). If YES in each determination step, a routine for setting design-related parameters, amplifier sure-related parameters, and delay-related parameters is executed (steps 121, 123, and 125). Furthermore, other switch processing is performed (step 126).

FIG. 28 shows the first interrupt routine using the aforementioned timer. First, XY coordinate data and pressure data are taken from the performance controls and stored in each register (step 127). Based on these stored data, the moving speed, direction and distance of the performance controls are calculated according to the routine described later and stored in each register. Store (step 128). Further, lateral vibration information is captured (step 129). Based on the above data, amplifier sure and design parameters are calculated according to a routine described later (step 130).

FIG. 29 shows the second interrupt routine using the aforementioned timer. First, delay length parameter processing is performed according to a routine to be described later (step 131). Next, calculate the loop gain of the sound source circuit, which will be described later (step 132).
Further, filter cutoff parameter processing, filter resonance parameter processing, and other parameter processing are performed (steps 133, 134, and 135).

FIG. 30 shows step 1 of the interrupt routine of FIG.
30 shows an example of the parameter processing routine for Ampsure and Design in No. 30. First, in step 136, register E
The amplifier sure device to be processed is determined based on the DEV number. The device number (DEVN) is, for example, O
is calculated from standard values or other parameters,
1 is the x coordinate (X) of the tablet, 2 is the Y coordinate (Y), 3
is the pressure of the tablet (PR), 4 is the speed of the tablet (
VEL), 5 indicates the distance @ (D I ST).

If the register EDEV for amplifier sure is not "0", first read the input data of the device indicated by EDEV and input it to BUF (step 137), then 1
．． The mBUF(7) value is converted into amplifier sure data using a method compatible with EDEV, and this is manually input to the EBUF and stored (step 138). Subsequently, in steps 139 and 140, processing regarding lateral sway is performed. Here, KSEF (
EN) is a flag indicating whether lateral vibration (KSH) is effective for the ENth parameter, and DEP (EN) is a flag indicating whether lateral vibration (KSH) is effective for the ENth parameter.
Indicates the depth of the effect when the lateral vibration is applied to the th parameter. As for the parameter numbers on the EN number, for example, 1 indicates amplifier sure, 2 indicates design, 3 indicates delay length, 4 indicates loop gain, 5 indicates filter cutoff, and 6 indicates filter resonance. Next, in step 141, the design register P
The number of DEV is determined. If 0, step 14
At step 6, it is determined whether the value of EBUF (steps 138 and 140) is greater than a predetermined threshold. If it is smaller than the threshold, it is ignored as noise, the number of times of processing is cleared to O, and the processing ends (step 147). If it is larger than the threshold value, calculate 2 to a predetermined operator (step 155), perform a predetermined operation based on this, and store the result in register PBUF.
(step 156). Then register TIM
E is rewritten (step 157). Determination step 14
If the number is 1 and the number is other than O, the input data of the device indicated by PDEV is read out, inputted into BUF, and stored (step 142). This BLIF value is converted into design data using a method compatible with PDEV, and is input and stored in PBUF (step 143). Subsequently, a design correction calculation routine to be described later is performed (step 145). Next, processing for lateral vibration similar to steps 139 and 140 described above is performed (steps 148 and 149). Ampsure data EBUF and breath pressure data PBU obtained above
F is sent to the sound source (step 15o).

On the other hand, when the amplifier sure device register EDEV is 0 in step 136, the routine shown in FIG. 31 is performed. First, the device is determined based on the number of the breath pressure device register PDEV (step 151). PDE
If V is also O, an error is displayed (step 15).
2). When the value is other than 0, read the input data of the device indicated by PDEV, input it to BUF, and store it (step 153). (7) BUF (71) value is PDE
It is converted into design data using a method compatible with V, and this is converted into PB.
Input and store in UF (step 154). Next, PBU
F is compared to a predetermined threshold (step 160) and is ignored as noise if it is less than the threshold. If it is larger than the threshold, a predetermined operator KIK is determined (step 161
), a predetermined calculation is performed based on this, and the result is manually input into register EBUF (step 162). Subsequently, the register TIME is rewritten (step 163), and then the same lateral vibration processing as in steps 139 and 140 described above is performed (steps 164 and 165). The data EBUF for amplifier sure and the data PBUF for design obtained as described above are sent to the sound source (step 16).
6).

FIG. 32 shows step 1 of the interrupt routine of FIG.
31 shows the delay length parameter processing routine in No. 31. First, the device is determined based on the number in the delay length register DDE■ (step 167). If it is 0, the key code is manually input to the CD into the delay length key code register TG (step 170). If other than 0, DDEV
The manual data of the device indicated by is read out and stored in the BUF (step 168). This BUF value is converted into key code data using a method compatible with DDEV and inputted to the TGKCD (step 169). Next, it is determined whether or not a lateral wobbling effect is to be applied to the delay length (step 171). If it does not sway, use the TG
The KCD data is inputted as is into the KBUF (step 172). If there is a lateral shake, a lateral shake correction calculation is performed and input into the KBUF (step 173). Next, the value of KBUF is converted into a delay length and inputted into the delay length register DBUP (step 174). The delay length data obtained as described above is sent to the sound source (step 175).

FIG. 33 shows the loop gain parameter processing routine in the interrupt routine (Figure 3429). First, in step 176, the number of the gain device register GDEV is determined. If it is 0, the standard gain 5TG1.5TG2 is set as the loop gain G1, G2 input to the sound source circuit (step 177). If it is other than 0, the input data of the device indicated by GDEV is read out, inputted to BUF, and stored (step 178). This BUF value is converted into a damping coefficient using a method compatible with GDEV, and these are designated as Gl and G2. Next, lateral fluctuation processing is performed on the loop gain (
Steps 180 and 181), the finally obtained loop gains G1 and G2 are sent to the sound source (Step 182)
.

FIG. 34 shows the calculation routine of step 128 in the constant cycle interrupt routine (FIG. 28) by the timer. In step 183, the amount of movement ΔX5ΔY in each direction is determined from the difference between the previous and current XY coordinate positions. Step 1
84, the distance from the reference position (Xo, Yo) is 111Lx, L
Find y. Based on these data, the speed VEL, rotation equivalent amount LOT, rotation direction DIR, and movement path 11iD of the tablet (performance controller) are determined by the predetermined calculations shown in the diagram.
Find IsT (step 185.186.187.1
89), after the calculation is completed, store the current position data X, Y in the register for the next calculation (step 190).
).

FIG. 35 shows a first example of the design correction calculation routine in step 145 of FIG. First, the EBUF calculated in the amplifier sure and design parameter processing routines (3430 and 31) described above is compared with a predetermined threshold (step 191). If it is smaller than the threshold, it is ignored as noise. If it is larger than the threshold, it is determined whether the number of times of processing has reached a predetermined value (step 192). If not, delay length key code register TGKCD
from the predetermined operator Bll B2. Find Ba and B4 and convert them into 1, 2. Ks, K4 (Step 1
93), this Bl, B2. B3. B= is the 38th
The straight line c that separates the sound generation area shown in the figure corresponds to bl+b2+cl""'l and c2-b2 in the graph, respectively. In a conograph, the straight lines s and c are respectively y-b
It is expressed as ++bzx+y-cl+c2x. If the input in the X direction is xIn and the input in the X direction is yIll, for example, y is made a straight line by calculation correction.

When entering the sound generation area between 0, set Xln to X and J+
b2x+ ((cl-bl)”(cz-bz)x) Y
Calculation is performed using In/ytn MAX as y.

At step 193, 8. 3. 5. When 4 is obtained, the number of processing times is incremented (step 195). Next, 1. A predetermined calculation is performed based on 2+, 3+, and 4, and is stored in the PBUF (step 196). The calculation in step 196 is performed with the above Xln being X, b1◆b, x+
((cl-bl)”(Cz-bz)X) yen/ y
This corresponds to an operation performed with anMAX as y.

On the other hand, if the predetermined set value has been reached in step 192, AI+A2+As, A4 are obtained from TGKCD and are converted into 1. 2. As Kl and K4 (step 194), the calculation of step 196 is performed. This A
l, ^2+ A3. A4 is the above-mentioned Bl, B2. Bs
, Similar to B4, the straight lines separating the sound production areas shown in Fig. 38, bl, bz, cl-bl in the graph of c.
, c2-bz, respectively.

FIG. 36 shows a second example of the design correction calculation routine.

Similar to the first example described above ('tS35 diagram), EBUF is compared with a predetermined threshold value (step 197), and the number of processing times is compared with a predetermined set value (step 198). , TIME are incremented (step 199).

Next, in step 200, i is set and the illustrated calculation is repeated four times to obtain Kl, K2 . 4 is calculated for Ks. On the other hand, when the number of processing times reaches the set value in the determination step 198, the TGKC
D to Kl, K2. 3. Find 4 in (Step 2
02). This is how I got it: 1. Based on 2°Kl and 4, PB is determined in the same manner as step 196 in FIG.
UF is calculated (step 201).

FIG. 37 shows an example of a sound source circuit for a wind instrument algorithm according to the present invention. The design signal and amplifier sure signal corrected as described above are each input to a subtracter 2 which serves as a circuit input section.
03 and adder 205 manually. The subtracter 203 subtracts the design signal from the input signal of the signal line L2, thereby outputting a differential pressure signal for displacing the reed of the mouthpiece. A low-pass filter 204 is connected to the output side of the subtracter 203 to remove high-frequency components of the differential pressure signal. This is because the lead does not respond to high frequency components. The adder 205 adds the amplifier sure signal and the output of the low-pass filter 204 to create a nonlinear table 2.
Output to 06. This nonlinear table 206 simulates the amount of displacement of the lead with respect to the applied pressure, and has predetermined input/output characteristics. As a result, the output of the nonlinear table 206 becomes a signal representing the air passage area in the reed of the mouthpiece. This nonlinear table 20
The output of 6 is connected to one power supply of multiplier 216. The differential pressure signal from the subtracter 203 is input to the other input side of the multiplier 216 via the nonlinear table 207. This nonlinear table 207 simulates that even if the differential pressure increases, the flow rate is saturated in a narrow pipe and the differential pressure and flow rate are not proportional. Based on these two input signals, the output signal of multiplier 216 is a signal representative of the airflow velocity in the mouthpiece reed.

Multiplier 216 is connected to adder 210 via attenuator 209. The attenuator 209 has the aforementioned calculation routine ('s
The loop gain G1 obtained in Figure 33) is manually input. This attenuator 209 is connected to the input end of an adder 210.

Adder 210 and adder 211 constitute a junction. Adder 210 combines the output side signal of delay circuit 215 and attenuator 209 to configure signal line L2.
The output signal is added to the output signal and output to the signal line L1. The other adder 211 adds the signal on the signal line L1 and the signal from the delay circuit 215 and outputs the result to the signal line L2. This loop simulates the combined pressure of the incident wave due to the human flow velocity and the reflected wave from the resonant tube immediately after the gap between the mouthpiece and the reed.

The signal on the signal line L1 is passed through a filter 213 and an attenuator 214.
and is fed back to the signal line L2 via the delay circuit 215. The filter 213 uses a low-pass filter alone or a combination of a low-pass filter and a bypass filter. The filter cutoff parameters and resonance parameters calculated in the aforementioned interrupt routine (FIG. 29) are input to the filters 204 and 213. Attenuator 2
14 is the loop gain G obtained by the calculation routine in Figure 33.
2 is input. The delay length parameter obtained in the calculation routine of FIG. 32 is input to the delay circuit 215. Filter 213 simulates the shape of a resonant tube. The delay circuit 215 simulates a state in which an incident wave from the mouthpiece returns to the mouthpiece as a reflected wave, corresponding to the length of the resonance tube and the length from the end of the resonance tube to the tone hole.

The waveform signal of the signal line L1 is extracted as an electronic sound output through a bandpass filter 212 for simulating the radiation characteristics of musical sounds in the air.

FIG. 38 is a relationship graph between design (pr) and amplifier sure (em). The shaded area between the straight line ■ and 0 indicates the range in which the sound is produced normally. The pen on the 3D tablet mentioned above is XY
In addition to coordinates, pen pressure is output. Therefore, for example, Ampsure can be calculated from (Y coordinate) x (pen pressure/maximum value of writing pressure), and design can be calculated from Ampsure.

Assume the equations of the lines ■ and ■ as follows.

■: pr=em/a+a ■: pr=em/d+d' In this case, the design pr in the shaded area is pr = em
/ d+ d'+ ((1/ a -17a) a
It can be obtained as m + a-d')X (X coordinate)/(maximum value of X coordinate).

In the wind instrument algorithm, pressure and X, Y using a 3D tablet (or mouse, joystick)
Examples of assignments between coordinate input parameters and musical tone control parameters are listed below.

B. Make the pressure correspond to Ampsure, and make the X coordinate correspond to the design.

The mouth and speed (calculated from the X and Y coordinates) correspond to Ampsure, and the pressure corresponds to the design.

C. Make the Y coordinate correspond to Ampsure, and make the pressure correspond to the design.

2. Adjust the pressure to the degree of rise and fall,
Looking for amp sure from speed, and looking for design instead. Make the sound rise when the pressure changes from 0. In this case, the relationship between the time at the rise of the sound and the volume is expressed as
Shown in Figures (a) and (b). (a) shows a case where the pressure change is large, and (b) shows a case where the pressure change is small. Execute when the speed is 0, such as at the start of an operation.

Conversely, when the pressure changes to 0, the sound disappears.

The relationship between volume and time in this case is shown in Figure 40 (a) and (b).
Shown below. (a) shows a case where the pressure change is large, and (b) shows a case where the pressure change is small.

Execute when the speed becomes 0, such as at the end of an operation.

E, adjust the pressure to the degree of rise and fall,
The distance from the center coordinates of the operator corresponds to Ampsure, and the speed corresponds to the design.

To, the pressure corresponds to the degree of rise and fall,
Find the amplifier sure from the distance from the center coordinates of the controller,
Furthermore, the design is calculated from Ampsure.

FIG. 41 is an explanatory diagram of a parameter control method when lateral vibration information is taken into consideration. Apply vibrato when lateral sway information is input 0 Normally, the design is controlled so that it moves on the dotted straight line (■) in the figure, where design = pressure x speed. In this case, the amp sure is calculated from the design by, for example, the amp sure value constant multiplied by the design. When the lateral vibration information is received, the design is fixed at that point and Ampsure is set to point G on the straight line.
, change within the range of the straight line ■■ according to the lateral sway information. This allows you to easily apply vibrato.

Note that in this example, the sound generation area characteristics of a bowed string instrument or a wind instrument are approximated by four straight lines, but this may be any number depending on the physical sound source algorithm used, or it may be a curved line. You can.

[Effects of the Invention] As explained above, in this invention, when controlling musical tone parameters based on operation information of performance operators,
In order to ensure that the electronic sound created using the musical sound parameters falls within a normal sound generation area, the musical sound parameters are input to the sound source after correction based on a sound generation area characteristic diagram based on the musical sound parameters. Therefore, regardless of the operation state of the performance operator, the sound is reliably produced, and the electronic musical instrument can be easily played.

By switching and controlling the onset and duration of a sound based on the sound generation area characteristic map, reliable pronunciation can be obtained and the range of sound expression can be expanded.

In particular, by using the sounding area characteristic diagram of the wind algorithm, it becomes possible to play the tones of a wind instrument with an electronic musical instrument. In this case, the power of expression will be richer than when playing only with the keyboard. It is easier to play than an actual wind instrument, and the player can breathe easily because the sound is reliably produced by operating the controls. When playing a wind instrument, the sound is adjusted by adjusting the fit and design of the reed, but with the controls, you can easily adjust the sound by simply moving your hand. Since you can move your hands freely on the controls, you can have a wide dynamic range.

You can easily control the volume and sound quality.

In addition, it is possible to intuitively match the movement for pronunciation with the sound. Furthermore, the duration of the sound can be extended.

[Brief explanation of drawings]

Fig. 1 is a basic configuration diagram of the electronic musical instrument control mechanism according to the present invention, Fig. 2 is a sound generation area characteristic diagram of the bowed string algorithm, Fig. FIG. 5 is a basic configuration diagram of the electronic musical instrument according to the present invention; FIG. 6 is a flowchart of the main routine of program control according to the method of the present invention; FIG. 7 is a mode diagram. Flowchart at the time of switching, Figure 8 is a top view of the slide volume type performance controller, Figures 349 (a) and (b) are side views and top views of the main parts of the performance controller in Figure 8, and Figure 10 is a top view of the slide volume type performance controller. Figure 11 is a flowchart when the key is turned on, Figure 12 is an explanatory diagram when selecting a tone, Figure 13 is an explanatory diagram of the channel register table, Figure 14 is a flowchart of the timer interrupt routine, Figure 15 is a flow diagram of the sound source control routine, Figure 16 is an explanatory diagram of the read management data table, Figure 17 is a configuration diagram of the sound source circuit of a bowed string instrument, Figure 18 is a sound generation area characteristic diagram of the wind algorithm, and Figure 19. Figure 20 is a flow diagram of the processing switching operation at the rise and sustain of the sound of an electronic wind instrument. Figure 21 is the block diagram of the sound source control mechanism of an electronic wind instrument using a 3D tablet. Figure 22 is a basic configuration diagram of an electronic wind instrument. Figure 23 is an explanatory diagram of the main flow of the sound source control program. Figure 24 is an explanatory diagram of the operation at key-on. Figure 25 is a flowchart when design-related device assignment is performed. , Fig. 26 is a flow diagram of the lateral wobbling effect, Fig. 27 is a flow diagram of panel switch processing, Fig. 28 is a flow diagram of the first example of the interrupt routine, and Fig. 29 is a flow diagram of the 542 example of the interrupt routine. Figures 30 and 31 are flowcharts of the amplifier sure and design parameter processing routines, Figure 32 is a flowchart of the delay length parameter processing routine, Figure 33 is a flowchart of the loop gain processing routine, and Figure 34 is a flowchart of the loop gain processing routine. Flowchart of the calculation processing routine, Figure 35 is a flowchart of the first example of the design correction routine, Figure 36 is a flowchart of the second example of the design correction routine, Figure 37 is a configuration diagram of the sound source of the wind instrument algorithm. , Figure 38 is a graph of the relationship between the design of the wind instrument algorithm and amplifier sure, Figures 39 (a) and (b) are graphs of the relationship between the time at the rise of the sound and the volume when the pressure is high and low, respectively. Figures (a) and (b) are graphs of the relationship between the time when the sound fades and the volume when the pressure is large and small, respectively, and Fig. 41 is an explanatory diagram of the control of the lateral vibration effect in the wind instrument algorithm. 1.15, 27, 74: Performance operator, 4: Correction circuit, 5
．． 13.78: IL version, 6.23: Sound source, 18:
(:PU. 76 Conversion program. Fig. Fig. Fig. Fig. 7 Routin Fig. Mo +7J conversion on event Fig. Fig. 3 Timer t11 - hours Fig. 4 Fig. 6 Fig. Figure 7 ) (b) Figure 41

Claims (2)

[Claims] (1) Control of a sound source for an electronic musical instrument characterized by correcting operation data of a performance operator corresponding to a musical tone parameter based on sound generation area characteristics corresponding to the musical instrument, and then inputting the corrected data to a sound source circuit. Method. (2) The above-mentioned sound generation area characteristics are represented by a graph consisting of four curves in a coordinate system with two musical tone parameters as the vertical and horizontal axes, and the area between the two central curves essentially constitutes the sound generation area. The electronic musical instrument according to claim 1, wherein the regions on both sides constitute a sound sustaining region, and the region outside the two outermost curves constitutes an irregular tone region. How to control the sound source.