JPH087586B2 - Electronic musical instrument sound source control method and electronic musical instrument - Google Patents

Description

BACKGROUND OF THE INVENTION 1. Field of the Invention The present invention relates to a sound source control method for an electronic musical instrument and an electronic musical instrument, and in particular, based on input data such as position and pressure corresponding to a musical tone parameter from a performance operator. The present invention relates to an improvement for a sound source circuit to always sound normally.

[Prior Art] Electronic devices that generate a player of a stringed instrument such as a violin or a wind instrument such as a clarinet are subject to mechanical vibration of the strings corresponding to the movement of the contact point between the strings and the bow or air vibration in the mouthpiece of the wind instrument. The sound source is provided with a physical sound source that generates an electronic sound that is physically approximated by an electric circuit. In such an electronic device, the key pitch information is input by operating the keyboard and the parameters corresponding to the bow pressure and the bow speed of the string operation or the breath pressure and the embouchure of the blowing operation are performed by the performance operator such as a slide volume. A control signal is input to the sound source to create and generate electronic sound.

In the conventional electronic musical instruments, the musical tone control signal depending on the operating position and the operating pressure of the performance operator is practically directly input to the sound source only by multiplying it by a certain coefficient regardless of the region such as speed and pressure.

[Problems to be Solved by the Invention] However, when the operation information of the performance operator is directly input to the sound source, there is no sound in a certain operation area, or an irregular sound such as an unpleasant sound or a so-called inside-out sound is generated. was there. Therefore, it is not easy to play the electronic musical instrument because it is necessary to operate the performance operator while avoiding such irregular sound generation.

The reason for the occurrence of such an irregular sound occurs, for example, in the case of a stringed musical instrument, because the parameter relation between the bow pressure and the bow speed is not within the normal sounding region. As shown in FIG. 2, the relationship between the bow pressure and the bow speed of a stringed instrument is as follows: a normal sounding area A in which a sound starts to sound and a sustaining area B in which a sound that has occurred once continues, with four straight lines passing through the origin. , 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 (v ₁ ), the bow pressure is too high or too low to enter the sounding area A, and no sound starts. Further, when the sound enters the irregular sound area C, the sound disappears, or an unpleasant sound or a flip sound is generated.

In the conventional electronic musical instrument, since the operation information of the performance operator is substantially directly input to the sound source, it may enter an irregular sound area depending on the operation state.In such a case, the sound disappears or an unpleasant sound is generated. Had occurred.

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 method of controlling a sound source for an electronic musical instrument and an electronic musical instrument, which can be always played in a normal sounding state regardless of an operating state of a performance operator. And

[Means for Solving the Problems] In order to achieve the above object, the invention according to claim 1
For the tone generator that generates a tone signal based on a plurality of types of parameters generated by the parameter generator, the parameters generated by the parameter generator are generated by the tone generator defined by the plurality of types of parameters. It is characterized in that the correction is performed based on the feasible region, and the corrected parameters are input.

According to a second aspect of the present invention, there is provided an electronic musical instrument having parameter generating means for generating a plurality of types of parameters and sound source means for generating a tone signal based on a plurality of types of parameters generated by the parameter generating means. Correction means for correcting a parameter generated from the parameter generating means to a parameter of a soundable region of the sound source means defined by the plurality of types of parameters, and inputting the corrected parameter to the sound source means. It is characterized by being provided.

Further, in the invention according to claim 3, the electronic musical instrument according to claim 2 is further provided with a sounding instructing means for instructing generation of a tone signal, and the correction means is (a) Within a predetermined time after the generation is instructed, the parameter generated from the parameter generating means is corrected to the parameter of the sounding area of the soundable area of the sound source means, and (b) the tone generating means is used. After a predetermined time from the generation of the signal, the parameter generated by the parameter generating means is corrected to a parameter of a continuous area other than the sound generating area of the sound generating means of the sound source means. To do.

[Operation] The parameter generated by the parameter generating means is corrected based on the soundable region of the sound source means defined by a plurality of types of parameters, and the corrected parameter is input to the sound source means.

Embodiment Hereinafter, the present invention will be described in more detail based on an embodiment with reference to the drawings.

FIG. 1 is a block diagram of an electronic stringed instrument according to the present invention. The performance operator 1 is composed of, for example, a slide volume provided with pressure sensitive means, a joystick mechanism, or a mouse mechanism. The position information by the operation of the operator 1 is converted into speed data (iv) through the A / D converter 2 and the speed conversion arithmetic circuit 3 and input to the correction circuit 4.
In addition, the pressure information from the pressure sensitive means of the operator 1 is the A / D converter 3
Is input to the correction circuit 4 as pressure data (ip).

The correction circuit 4 corrects the velocity data (iv) and the pressure data (ip) at the rising time which is the first sound generation so that the speed data (iv) and the pressure data (ip) enters the sound generation area where the sound starts to sound, and the bow speed data (vv) and the bow data. Input to the sound source 6 as pressure data (vp). Further, the pitch information (p) corresponding to the scale is input to the sound source 6 by operating the keyboard 5. The sound source 6 creates an electronic sound based on the bow speed data, bow pressure data and pitch data, and emits the sound through the sound system 7.

A calculation method for correcting the tone generation area in the correction circuit 4 will be described below.

Bow speed vv and bow pressure, which are parameters for controlling the sound of a stringed instrument
The relationship of vp is shown in FIG. 2 as described above. The four straight lines a, b, c, d pass through the origin, and the inclination differs depending on the pitch information from the keyboard. Therefore, the correction calculation program is executed based on the table of key numbers. Also,
Each straight line also depends on the distance from the top of the plucked string instrument to the plucked position. Therefore, the calculation program includes a table having the distance from the frame as a parameter. The bow speed has positive and negative values, but the graph is omitted because it is symmetric with respect to the vertical axis (bow pressure).

Bow pressure vp to ensure normal pronunciation regardless of bow speed
An example of a calculation formula for correcting and positioning in the area A is shown below.

vv = iv ... vp = {(c + b) / 2} × iv ... By such a calculation, if vv is not zero, the sound always enters the sounding area A and a sound starts to sound. In this case, the volume changes depending on vv, but the sound quality is constant and monotonous. Also, the pressure data (ip) is ignored. In addition, c and b represent the inclination of each straight line c and b.

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

vp = b × iv + (c−b) × iv × ip / P _max ... The above expression changes the sound quality by changing the position in the vertical direction in the sound generation area A based on the pressure data from the operator.
P _max is the maximum value of the input pressure data (ip).

The physical sound source that generates the electronic sound has a hysteresis characteristic that, even after the sound is generated once, the sound is normally sounded in the sustaining area B outside the sounding area A. Therefore,
After the sound begins to sound once, it is desirable that the position of the parameter be widely changed including the continuous region to have a wide range of sound quality.

A calculation formula for expanding the range of such a sound generation region is shown below.

vp = a × iv + (d−a) × iv × ip / P _max ... By controlling the sound source by the formula and the formula, it is possible to normally produce sound in a wide range and increase the change in the sound volume.

FIG. 3 shows a flow for switching the continuous processing in which the control area is expanded to the continuous area B and the rising processing for starting the sound.

The number of times the rising process is repeated from the rising is set in advance. At step 301, it is judged if the counter has reached this set value. If it is less than the set value, the rising process is repeated (step 302). After processing, the counter is incremented and incremented by 1 (step 30
4) Return to step 301. When the counter reaches the set value, continuous processing is performed (step 303). Once the sustaining process is performed, the sustaining process is repeated through the determination step 301 by the counter increment (step 304). 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 correction calculation described above, ip and iv are not used as they are for inputting the expression, but a table based on key numbers and the like is passed through, so that it is possible to control musical tone parameters in accordance with human senses.

Further, in the above correction calculation, ip was created from iv, but conversely, iv can be created from ip. As an example of such a calculation, the following equations are shown corresponding to each of the above equations.

vp ＝ vi ・ ・ ・ ・ ・ ・ vv ＝ (1 / d) × ip ＋ {(1 / a) − (1 / d)} × ip × iv / V _max .... where V _max is the maximum of iv It is a value. Each arithmetic expression can be selected according to the playing feeling according to the musical instrument. By adjusting the slopes a, b, c, d of the straight lines a, b, c, d of the conversion characteristic graph shown in FIG. 2, the range of each region is adjusted to include the irregular sound region C slightly, etc. Can widen the range of performance expression. In this case, the conversion characteristic may be set when playing.

FIG. 4 shows an example of a graph of characteristics for the pronunciation correction calculation of the rubbing string algorithm of an electronic string instrument using a three-dimensional tablet as an input device. A range between the straight lines a and c is a sounding region A where a sound starts to sound, a continuous region B where the sound continues between the straight lines b and d, and an irregular sound region C is outside the straight line bd. The slopes a, b, c, and d of each straight line are rubbing points (distance from coma)
It depends on It also changes depending on the key number. In particular, d changes greatly depending on the key number. In such a characteristic graph, the sounding area A
If it is not in, the sound will not start. The following is a calculation formula for correcting the bow speed from the bow pressure and always entering the sound generation area A at the time of rising.

vb = fb / c + {( 1 / a) - (1 / c)} × vb / V max × fb , where, vb, fb denotes each bow speed and Yumi圧. After the sound starts to be heard, the continuous area B is entered. The range below the straight line b is a region where the sound disappears, and the range above the straight line d is a region where an unpleasant sound is emitted. The following is a calculation formula for correcting the bow speed from the bow pressure and always entering the sustaining area B after rising.

vb = fb / d + {( 1 / b) - (1 / d)} × vb / V max × fb FIG. 5 is a block diagram of a control mechanism of the electronic musical instrument having the above-described correction circuit.

Signals from the performance controller 15 and keyboard 13 are detected by the detection circuit.
CP from bus line via 16 and keyboard switch circuit 14
Input to U18. The CPU 18 reads out the necessary data from the program ROM 19 storing each routine program, the data ROM 20 storing the data necessary for the arithmetic processing, and the work RAM 21 storing each calculation result during the arithmetic processing and performs the above-described correction arithmetic operation. The calculated tone control parameter is calculated.
The function operator 22 normally selects a tone color, vibrato, etc. and switches various modes. For example, the bow position detection mode and the bow speed detection mode are switched. Timer 17
Several ms for the main routine of the program by CPU18
Performs an interrupt routine at a fixed cycle.

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

FIG. 7 shows a mode switching routine. In step 11, the mode such as the detection mode is switched, and the detection result is stored in the register for the next detection calculation process (step 12).

An example of an input device for controlling a musical tone parameter 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 an essential part, respectively. This input device is a slide volume type operator, and an operator 27 slides along a central guide groove 26 of a main body 25 which constitutes the first slide volume. As shown in FIG. 9, the operator 27 includes a slider 28 that slides along the guide groove 26 as shown by an arrow D, and the slider 28.
It consists of an operating piece 29 mounted on top. The operation piece 29 is preferably rotatable with respect to the slider 28 as indicated by arrow F in order to facilitate the sliding operation. In this case, the rotation angle can be detected and used as the musical tone control data. The operation piece 29 constitutes a second slide volume. Along the guide groove 30 of the operation piece 29, the slider 31 moves in the direction of arrow E.
Slide like. The first position data is obtained from the resistance value according to the position of the manipulator 27, and the second position data is obtained from the resistance value according to the position of the manipulator 31.

A pressure sensor 32 is attached to the side surface of the manipulator 29, and the pressure during operation is measured to obtain pressure data. These two
The tone control parameters are calculated based on the one position data and the pressure data, and the above-described correction calculation is performed.

FIG. 10 shows a key-on routine by the CPU. First, the key code of the depressed key is stored in the key code register (KCD) (step 33). Next, the sound generation channel of the sound source is assigned. The assigned channel is stored in the assign channel register (ACH) (step 34). Next, from the specified read control data (TCD) to KCD
The filter coefficient of the tone control filter circuit, which will be described later, corresponding to the key code is read and sent to the assigned channel (ACH) of the sound source (step 35). Next, the assigned channel is instructed to be sounded and registered (step 36). At this time, input the signal "1" to the flag of the registered channel.

FIG. 11 shows a routine at the time of key off. First, the key code of the released key is stored in the KCD (step 3
7). Next, the sound channel of the sound source to which the key code is assigned is searched using the channel table (step 38). In the determination step 39, it is determined whether or not there is such a channel. If not, exit the routine,
If so, it is judged whether or not all other channels are "0" (step 40). If the flags of the other channels are all "0", the routine is terminated, and if not "0", "0" is input to the flag of the assigned channel (step 41).
Next, the release attenuation coefficient of the sound corresponding to the key release key code is read from the read table (step 42). This attenuation coefficient is sent to the assigned channel (step 43). Next, the number of times of processing of the assigned channel is cleared to "0" and the routine ends (step 44).

FIG. 12 shows a tone color selection routine. A tone color number transmitted from a tone color selection operator (for example, a selection switch on the instrument body or the above-mentioned performance operator) is input to a predetermined register (step 45).

FIG. 13 shows an example of a register table for managing the sound generation channels of the sound source. In this example, the number of sound source channels is four, which corresponds to the number of strings of the violin. By providing a plurality of strings in this way, the reverberation effect of the original sound source can be obtained when the key-on signal is transferred from one channel to another channel. When checking the channel flag in each routine, the number i is set, i is incremented by 1 for each check process from the first number 0, and it is repeated 4 times until it becomes 3.

FIG. 14 shows an interrupt routine that interrupts the main routine at fixed intervals with a fixed clock. First, the slide volume type performance operator described above (see FIGS. 8 and 9).
The first and second position data and the pressure data are stored in the predetermined registers POS1, POS2, PRES by the operation shown in the figure) (step 46). Next, in step 701, mode discrimination is performed. If the mode is "1", the bow velocity v is directly obtained from the first position data (POS1) of the manipulator using the conversion table (PVTBL) created and stored in advance (step 47).

If the mode is “0” in step 701, the speed is calculated from the difference between the first position data of the previous time and this time and the register (VE
(L) (step 48). At this time, since the detection timing is constant, the position difference directly corresponds to the speed. Next, the velocity data VEL is converted into the bow velocity v by using another conversion table (VVTBL) created and stored in advance (step 49). If the bow speed v is obtained, this is the first
The position data POS1 of 1 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 value (step 51). If it is smaller than the threshold value, it is ignored as noise and "0" is input to the number of processing times of each channel (step 52). When the bow speed v is greater than the threshold value, the sound source control routine is executed using this bow speed v (step
53), each parameter of the sound source is calculated based on the input data as described later, and is sent to the sound source.

FIG. 15 shows a sound source control routine. First, the process is executed for the channel "0" in the channel table (FIG. 13) described above (step 54). The flag for this channel is "1"
It is determined whether or not it is a channel to be controlled (step 55). If not "1", the determination is repeated for the other three channels (step 66). If the flag is "1", the key code of this channel is stored in the KCD register (step 56). Next, the tone number (12th
Attenuation coefficient data and delay length data corresponding to the key code are obtained from the tone color data group (see the figure) (step
57). Next, a predetermined operation is performed using the second position data POS2 and the result is stored in the register PO (step 58). Next, it is judged whether or not the number of rising processes has reached a predetermined set point TMAX (step 59). If not reached, the straight line inclinations b, c-b corresponding to the key code are read from the tone color data group in order to put the control parameter in the tone generation region A (see FIG. 2) of the tone color (sounding) characteristic, and in the figure. A predetermined calculation for obtaining C1 and C2 shown in (1) is performed (step 61). When this calculation is completed, the processing number i is incremented (step 62).

On the other hand, when the number of times of processing reaches the predetermined set value in the judgment step 59, both continuous regions B of the sounding characteristic (see FIG. 2).
In order to put the control parameter in the range of, the slopes a and da of the straight line corresponding to the key code are read from the tone color data group, and a predetermined calculation for obtaining C1 and C2 shown in the figure is performed (step 60). ).

Next, the bow pressure P is calculated based on the above C1 and C2 (step 63). Further, data D1 and D2 for controlling two delay circuits of the tone generator circuit described later are calculated (step 6).
Four). These D1 and D2 are DC (two damping coefficients), P (arch pressure),
It is sent to the processing channel of the sound source together with v (bow speed) (step 65). In the figure, DD1 and DD2 each represent a delay reference length of two delay circuits, and ΔDD represents a change width of the bow position. The above process is repeated for four channels (step 66).

FIG. 16 shows an example of a register table storing the above-mentioned tone color data group. 67 indicates a read management data area, which has the number of areas corresponding to the number of key codes. In each management data area, as shown in the drawing, the attenuation coefficient data register 68, the release attenuation coefficient data register 69, the filter coefficient data register 70, the delay length data register 7
1. The data register 72 for vp conversion at the time of rising and the data register 73 for vp conversion at the time of continuation are stored.

FIG. 17 shows an example of a physical sound source circuit for electronic sound generation of a stringed instrument. Reference numerals 702 and 703 denote adders corresponding to rubbing chord points.
Reference numerals 704 and 705 denote multipliers, which correspond to the chord ends on both sides of the rubbing chord point.
The closed loop including the adder 702, the delay circuit 706, the low-pass filter 707, the attenuator 708, and the multiplier 704 corresponds to the chord on one side of the critical chord point, and the delay time of the closed loop corresponds to the resonance frequency of the chord. Similarly, the closed loop including the adder 703, the delay circuit 709, the low-pass filter 710, the attenuator 711, and the multiplier 705 corresponds to the chord on the other side of the chord point.
Reference numeral 712 represents a non-linear function generator. In this non-linear function generator 712, a signal corresponding to the bow velocity is added to the signal obtained by combining the outputs of the closed loops on both sides of the rubbing chord point in the adder 713, and further, the fixed hysteresis low-pass filter 71.
A signal obtained by adding a signal obtained by multiplying the signal from 4 to a gain G by a multiplier is input. In addition, the non-linear function generator 712
The hysteresis control of is performed by the signal corresponding to the bow pressure.

Next, an embodiment in which the present invention is applied to control of a sound source for an electronic wind instrument will be described. In the wind instrument algorithm, the pronunciation characteristic according to the relationship between the embouchure and the breath pressure is the 18th, which corresponds to the pronunciation characteristic diagram (FIG. 2) of the above-mentioned stringed instrument.
It is approximated as shown. Unlike the case of a rubbing instrument, the four straight lines do not pass through the origin and each has a segment. As in the case of the stringed instrument, the range between the two central straight lines b and c is the sounding area A, and the range on both outer sides is the continuous area, and the outside of the two outer straight lines a and d. The range is the irregular sound area C. Further, the inclination of each straight line varies greatly depending on the pitch.

FIG. 19 shows an example of a control system for correcting a control signal from a performance operator based on such a sounding region characteristic of a wind instrument and inputting the corrected signal to a sound source. As the input device, a performance operator 74 composed of a three-dimensional tablet that draws a locus on the XY plane is used. The pen pressure data for gripping the performance operator and the XY coordinate data indicating the X position and the Y position are input to the sounding area correction program 76 of the detection circuit 75. The input data is converted by the conversion program 76 into breath pressure data and embouchure data having a value that falls within a predetermined normal sounding region,
Input to the sound source control circuit 77.

On the other hand, by operating the keyboard 78, the key number is input to the sound source control circuit 77 as pitch information.

Similar to the case of a stringed instrument, a physical sound source that generates an electronic sound has a hysteresis characteristic that sustains a normal sound generation in a sustaining area B outside the sounding area A after the sound is once generated. Therefore, it is desirable to change the position of the parameter widely including the continuous region to give a wide range of sound quality once the sound starts to sound.

FIG. 20 shows a flow for determining whether to switch between rising processing for starting sound and continuous processing after rising. First, the maximum value of the embouchure is set (step 7
9). Next, the inclinations and intercepts of the four straight lines a, b, c, d (Fig. 18) are read from a predetermined table (step 80). Next, the intercepts for rising and for addition are calculated (step 81).
Next, at step 82, it is judged if it is a rising process.
This determination is performed, for example, by counting the number of rising processes and determining whether the counted number has reached a predetermined set value.
If it is the rising process, the parameter is determined by the key code (step 83), the minimum value of the embouchure is set (step 84), the breath pressure at the rising time is calculated (step 85), and the breath pressure is corrected. (Step 86). When this breath pressure correction is completed, the count is incremented (step 87). When the number of counts exceeds a predetermined set value, the process is switched to continuous processing, the parameter by the key code is determined (step 88), the minimum value of the embouchure is set (step 89), and the breath pressure for sustaining the sound is set. It is calculated (step 90) and the breath pressure is corrected (step 91).

FIG. 21 is a block diagram of a sound source control system for a wind instrument type electronic musical instrument according to the present invention. X-position data, Y-position data and writing pressure data are transmitted from the performance operator 74, which is a three-dimensional tablet, and stored in the register 94. The X, Y coordinate data is calculated in speed, direction and distance by the arithmetic circuit 93 at regular time intervals using the reference coordinates X ₀ , Y ₀ stored in the reference point coordinate register 92, and the calculation result is stored in the register 94. To be done. The register 94 is connected to the tone control parameter calculation circuit 95.

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

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

FIG. 22 is a block diagram of a 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 bus line. The CPU 18 calculates the tone control parameters by reading out the necessary data from the ROM 103 storing the routine programs and the data necessary for the arithmetic processing and the RAM 104 storing the calculation results during the arithmetic processing. The panel switch 105 is used to select a tone color, vibrato, etc., and switch various modes. The display 106 displays the selected switch or mode. Timer 17
Is for performing an interrupt routine with respect to the main routine by the CPU 18 at a fixed cycle of about several ms.

FIG. 23 shows a basic main routine. Step 107
At, each arithmetic circuit is initialized, and each sound source parameter is set to a predetermined initial value. Subsequently, the key switch processing (step 108) and other switch processing (step 109) of the keyboard are repeated. With respect to such a main routine, an interrupt routine (described later) is executed at a constant cycle by the timer to calculate the various control parameters.

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 depressed key is stored in the register KCD (step 110).

FIG. 25 shows the routine when the numerical input to the breath pressure control device for setting the parameter related to the breath pressure in step 109 of the main routine is turned on. First, the input numerical value is stored in the register BUF (step 11).
1). The data in BUF is stored in the breath pressure device register PDEV (step 112). Subsequently, the breath pressure control device name is displayed (step 113). In this embodiment, a device (operation data) for controlling breath pressure or the like can be arbitrarily selected.

FIG. 26 shows the routine for the lateral shake effect switch-on event in step 109 of the main routine. Step
At 114, a flag indicating whether or not the horizontal shake effect is applied to a predetermined parameter is switched. In step 115, it is judged whether or not this flag is "1", that is, whether or not the horizontal shake is effective. If it is "1", horizontal shake on is displayed (step 117), and if it is not "1", horizontal shake off is displayed (step 116). In this embodiment, it is possible to select, for each parameter, whether or not to add the parameter due to the lateral wobbling of the keyboard.

FIG. 27 shows a panel switch processing routine in step 109 of the main routine. When the switch is turned on, the edit screen displaying the parameter to be processed is selected and the screen number is stored in the register PAGE (step 11).
8). The screen of this stored number is displayed (step 119). Subsequently, it is sequentially determined whether or not the breath pressure, the embouchure, the delay, and other editing processes are performed (determination steps 120, 122, and 124). If YES in each determination step, a routine for setting a breath pressure related parameter, an embouchure related parameter, and a delay related parameter is executed (steps 121, 123, 12).
Five). Further, other switch processing is performed (step 126).

FIG. 28 shows the first interrupt routine by the timer described above. First, XY coordinate data and pressure data are fetched from the performance operator and stored in each register (step
127). Based on these stored data, the moving speed, direction and distance of the performance operator are calculated according to the routine described later and stored in the registers (step 128). Further, the horizontal shake information is captured (step 129). Each parameter of the embouchure and breath pressure is calculated based on the above-mentioned data according to a routine described later (step 130).

FIG. 29 shows a second interrupt routine by the timer described above. First, a delay length parameter process is performed according to a routine described later (step 131). Subsequently, a loop gain of the tone generator circuit described later is calculated (step 132), and further filter cutoff parameter processing, filter resonance parameter processing, and other parameter processing are performed (steps 133, 134, 135).

FIG. 30 shows step 130 of the interrupt routine of FIG.
2 shows an example of a parameter processing routine for embouchure and breath pressure in FIG. First, at step 136, an embouchure device to be processed is identified by the number of the register EDEV. The device number (DEVN) is, for example, 0 for a standard value or calculated from other parameters, 1 for the X coordinate (X) of the tablet, 2 for the Y coordinate (Y), 3 for the pressure of the tablet (PR), 4 indicates the speed of the tablet (VEL) and 5 indicates the distance (DIST). When the register EDEV for embouchure is not "0", first the input data of the device indicated by EDEV is read and this is input to BUF (step 137).
Next, the value of BUF is converted into embouchure data by a method corresponding to EDEV, and this is input to EBUF and stored (step 13
8). Subsequently, in steps 139 and 140, processing for lateral shake is performed. Here, KSEF (EN) is a flag that indicates whether lateral shake (KSH) is effective for the ENth parameter, and DEP
(EN) indicates the depth of the effect when lateral shake is effective for the ENth parameter. As the parameter number of EN side, for example, 1 is embouchure, 2 is breath pressure, 3 is delay length, 4
Indicates a loop gain, 5 indicates a filter cutoff, and 6 indicates a filter resonance. Next, at step 141, the number of the breath pressure register PDEV is determined. If 0, step 14
At 6, it is determined whether the value of EBUF (steps 138, 140) is larger than a predetermined threshold value. If it is smaller than the threshold value, it is ignored as noise, the number of times of processing is cleared to 0, and the processing ends (step 147). If it is larger than the threshold, the predetermined operator K ₁ ,
K ₂ is calculated (step 155), a predetermined operation is performed based on this, and the result is input to the register PBUF (step 15).
6). Then rewrite the register TIME (step 15
7). If the number is not 0 in the judgment step 141, the input data of the device indicated by PDEV is read and this is input to the BUF and stored (step 142). The value of this BUF is converted into breath pressure data by a method corresponding to PDEV, and is input and stored in PBUF (step 143). Subsequently, a breath pressure correction calculation routine described later is performed (step 145). Then step 13 above
The same processing for lateral shake as in steps 9 and 140 is performed (steps 148 and 149). The embouchure data EBUF and breath pressure data PBUF obtained as described above are sent to the sound source (step 150).

On the other hand, when the embsure device register EDEV is 0 in step 136, the routine shown in FIG. 31 is executed.
First, the device is identified by the number of the breath pressure device register PDEV (step 151). When PDEV is also 0, it is displayed as an error (step 152). If the value is other than 0, the input data of the device indicated by PDEV is read and input to BUF for storage (step 153). The value of this BUF is converted into breath pressure data by the method corresponding to PDEV,
Input to PBUF and store (step 154). Next, PBUF is compared with a predetermined threshold value (step 160), and if smaller than the threshold value, it is ignored as noise. If it is larger than the threshold value, predetermined operators K ₁ and K ₂ are obtained (step 161), a predetermined operation is performed based on this, and the result is input to the register EBUF (step 162). Then, the register TIME is rewritten (step 163). Next, the same processing as that of the above-described steps 139 and 140 is performed (steps 164 and 16).
5), data E for Embussure obtained as above
The BUF and the data PBUF for breath pressure are sent to the sound source (step 166).

FIG. 32 shows step 131 of the interrupt routine shown in FIG.
3 shows a delay length parameter processing routine in FIG. First, the device is identified by the number of the delay length register DDEV (step 167). If it is 0, the key code is input to the delay length key code register TGKCD (step 1
70). If the value is other than 0, the input data of the device indicated by DDEV is read and input to BUF for storage (step 1
68). The value of this BUF is converted into key code data by a method corresponding to DDEV and input to TGKCD (step 169).
Next, it is judged whether or not the horizontal shaking effect is applied to the delay length (step 171). If there is no horizontal shake, the TGKCD data is directly input to KBUF (step 172). If there is horizontal shake, horizontal shake correction calculation is performed and this is input to KBUF (step 173). Next, the value of KBUF is converted into a delay length and this is input to the delay length register DBUF (step 174). The delay length data obtained as described above is sent to the sound source (step 17
Five).

FIG. 33 shows a routine for loop gain parameter processing in the interrupt routine (FIG. 29). First step 17
Determine the number of gain device register GDEV with 6. 0
If so, the standard gains STG1 and STG2 are set as loop gains G1 and G2 to be input to the tone generator circuit (step 177). If it is not 0, the input data of the device indicated by GDEV is read out and input to BUF for storage (step 178). This BUF value is converted into an attenuation coefficient by a method corresponding to GDEV, and this is converted to G1,
G2. Next, a horizontal fluctuation process is performed on the loop gain (steps 180 and 181), and the finally obtained loop gains G1 and G2 are sent to the sound source (step 182).

FIG. 34 shows the arithmetic routine of step 128 in the timer interrupt routine (FIG. 28) of a constant cycle. In step 183, the movement amounts ΔX and ΔY in each direction are obtained from the difference between the XY coordinate positions of the previous time and this time. In step 184, the distances L _X and L _Y from the reference position (X ₀ , Y ₀ ) are obtained. Based on these data, the speed VEL of the tablet (performance operator), the equivalent rotation amount LOT, the rotation direction DI
R, moving distance DIST (steps 185, 186, 187, 18)
9). After the calculation is completed, the position data X and Y of this time are stored in the register for the next calculation (step 190).

FIG. 35 shows a first example of the breath pressure correction calculation routine in step 145 of FIG. First, the parameter processing routine for the above-mentioned embouchure and breath pressure (Figs. 30 and 31)
The EBUF calculated in the figure) is compared with a predetermined threshold value (step 191). If it is smaller than the threshold value, it is ignored as noise.
If it is larger than the threshold, it is judged whether or not the number of times of processing has reached a predetermined value (step 192). If not reached, the predetermined operator B ₁ , from the delay length key code register TGKCD
B ₂ , B ₃ , B ₄ are obtained and set as K ₁ , K ₂ , K ₃ , K ₄ (step 19
3). The _{B 1, B 2, B 3} , B 4 are each corresponding to _{b 1, b 2, c 1} -b 1, c 2 -b 2 in the linear b, graph c to separate the sound area shown in FIG. 38 To do. In this graph, the straight lines b and c are y = b ₁
It is represented by + b ₂ x, y = c ₁ + c ₂ x. Input in the x direction is x _in , y
If the direction input is y _in , for example, y is a straight line calculated
To put it in the pronunciation area between b and c, set x _in to x and b ₁ + b ₂
x {(c ₁ −b ₁ ) + (c ₂ −b ₂ ) x} y _in / y _in Perform the operation with MAX as y.

When K ₁ , K ₂ , K ₃ and K ₄ are obtained in step 193, the number of times of processing is incremented (step 195). Next, this K ₁ , K ₂ , K ₃ , K ₄
Based on the above, perform a predetermined calculation and store in PBUF (step
196). Calculation of step 196 is the x _in the x b ₁ + b
₂ x {(c ₁ −b ₁ ) + (c ₂ −b ₂ ) x} y _in / y _in corresponds to the operation performed with MAX as y.

On the other hand, when the predetermined set value is reached in step 192, A ₁ , A ₂ , A ₃ and A ₄ are obtained from TGKCD and these are K ₁ , K ₂ , K ₃ and K ₄
(Step 194), the calculation of step 196 is performed. The _{A 1, A 2, A 3} , A 4 , similar to the _{B 1, B 2, B 3} , B 4, 38
In the graph of straight line b, c that divides the pronunciation area shown in the figure, b ₁ ,
It corresponds to b ₂ , c ₁ −b ₁ and c ₂ −b ₂ , respectively.

FIG. 36 shows a second example of the breath pressure correction calculation routine.
Similar to the above-mentioned first example (FIG. 35), 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). If it is less than the set value, T
Increment the IME (step 199). Next, at step 200, i is set and the illustrated operation is repeated four times to calculate K ₁ , K ₂ , K ₃ and K ₄ . On the other hand, if the number of processing times is the set value in the discrimination step 198, K ₁ , K ₂ , K ₃ and K ₄ are obtained from TGKCD in the same manner as in step 194 of FIG.
2). Based on K ₁ , K ₂ , K ₃ and K ₄ obtained in this way,
PBUF is calculated in the same manner as step 196 in the figure (step 201).

FIG. 37 shows an example of the tone generator circuit of the wind instrument algorithm according to the present invention. The breath pressure signal and the embouchure signal corrected as described above are respectively applied to the subtracter 20 which serves as a circuit input section.
3 and input to the adder 205. The subtractor 203 subtracts the breath pressure signal from the input signal of the signal line L2 to output a differential pressure signal for displacing the lead of the mouthpiece. A low-pass filter 204 is connected to an output side of the subtractor 203, and removes a high-frequency component of the differential pressure signal. this is,
This is because the reed does not respond to high frequency components. Adder 205
Outputs the sum of the embouchure signal and the output of the low-pass filter 204 to the non-linear table 206. The non-linear table 206 simulates the amount of displacement of the lead with respect to the applied pressure and has a predetermined input / output characteristic. As a result, the output of the non-linear table 206 becomes a signal representing the air passage area in the lead of the mouthpiece.
The output of this non-linear table 206 is connected to one input of a multiplier 216. The subtractor 203 is provided on the other input side of the multiplier 216.
The differential pressure signal from is input via the non-linear table 207. This non-linear table 207 simulates that even if the differential pressure becomes large, the flow velocity is saturated in a narrow pipeline and the differential pressure and the flow velocity are not proportional. Based on these two input signals, the output signal of the multiplier 216 becomes a signal representing the air flow velocity at the lead of the mouthpiece.

The multiplier 216 is connected to the adder 210 via the subtractor 209. The loop gain G1 obtained in the above-described calculation routine (FIG. 33) is input to the attenuator 209. The attenuator 209 is connected to the input side of the adder 210.

The adder 210 forms a junction with the adder 211. The adder 210 adds the signal on the output side of the delay circuit 215 for forming the signal line L2 and the output signal of the attenuator 209 and outputs the result 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 and the reflected wave from the resonance tube due to the input flow velocity immediately after the gap between the mouthpiece and the lead.

The signal on the signal line L1 is fed back to the signal line L2 via the filter 213, the attenuator 214 and the delay circuit 215. The filter 213 is a low-pass filter alone or a combination of a low-pass filter and a high-pass filter. filter
The filter cutoff parameters and resonance parameters calculated in the interrupt routine (FIG. 29) described above are input to 204 and 213. The loop gain G2 obtained by the calculation routine of FIG. 33 is input to the attenuator 214. Delay circuit 21
The delay length parameter obtained by the calculation routine of FIG. 32 is input to 5. The filter 213 simulates the shape of a resonance tube. The delay circuit 215 simulates a state in which an incident wave from the mouthpiece returns to the mouthpiece as a reflected wave according 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 taken out as an electronic sound output through the bandpass filter 212 for simulating the radiation characteristic of the musical sound in the air.

FIG. 38 is a graph showing the relationship between breath pressure (pr) and embouchure (em). The straight line and the shaded area between them indicate the range in which the sound normally sounds. The pen on the above-mentioned three-dimensional tablet outputs writing pressure in addition to XY coordinates. Therefore, for example, the embouchure can be calculated from (Y coordinate) × (writing pressure / maximum writing pressure), and the breath pressure can be calculated from the embouchure.

 The equation of the straight line, is assumed as follows.

: pr = em / a + a ': pr = em / d + d' In this case, the breath pressure pr in the shaded area is pr = em / d + d '+
It can be obtained as {(1 / a−1 / d) em + a′−d ′} × (X coordinate) / (maximum X coordinate).

In the wind instrument algorithm, pressure and X by a three-dimensional tablet (or mouse, joystick),
Listed below are examples of assignments of input parameters of the Y coordinate and tone control parameters.

I. The pressure corresponds to the embouchure and the X coordinate corresponds to the breath pressure.

B. Velocity (calculated from X, Y coordinates) corresponds to Embouchure and pressure corresponds to breath pressure.

C. Correspond the Y coordinate to the embouchure and the pressure to the breath pressure.

D. Corresponding the pressure to the degree of rising and disappearing,
The embouchure is calculated from the velocity and the breath pressure is calculated. Make the sound rise when the pressure changes from 0. In this case, the relationship between the rising time of the sound and the volume is shown in FIGS. 39 (a) and 39 (b). (A) shows the case where the pressure change is large, and (b) shows the case where the pressure change is small. It is executed when the speed is 0, such as when the operation is started.

On the contrary, the sound should disappear when the pressure changes to 0. The relationship between volume and time in this case is shown in Fig. 40 (a),
It shows in (b). (A) is when the pressure change is large, (b)
Indicates the case where the pressure change is small. It is executed when the speed becomes 0, such as at the end of the operation.

E. Corresponding the pressure to the degree of rising and disappearing,
The distance from the center coordinate of the manipulator corresponds to the embouchure, and the velocity corresponds to the breath pressure.

F. Corresponding the pressure to the degree of rising and disappearing,
Find the embouchure from the distance from the center coordinates of the manipulator,
Furthermore, the breath pressure is calculated from the embouchure.

FIG. 41 is an explanatory diagram of a parameter control method in consideration of lateral shake information. Vibrato is applied when lateral shake information is input. In normal times, breath pressure = pressure × velocity, and control is performed so as to move along the dotted straight line in the figure. In this case, the embouchure is calculated from the breath pressure, for example, with Ambushua = constant × breath pressure. When the lateral shake information is entered, the breath pressure is fixed at that point and the embouchure is moved to the point G on the straight line.
Then, it is changed within the range of the straight line according to the lateral shake information. This makes it possible to easily apply vibrato.

In this embodiment, the sounding region characteristics of a stringed instrument or a wind instrument are approximated by four straight lines, but this may be an arbitrary number depending on the physical sound source algorithm used, or a curved line. May be.

[Effects of the Invention] As described above, in the invention described in claims 1 and 2, when the parameter generated from the parameter generating means is sent to the sound source means, the electronic signal generated based on the parameter is transmitted. The tone parameters are input to the sound source after the parameters are corrected based on the soundable region of the sound source means defined by a plurality of types of parameters so that the sound always enters the normal sounding region.
Therefore, the sound is surely generated regardless of the operation state of the performance operator, and the electronic musical instrument can be easily played.

According to the third aspect of the invention, the region of the parameter to be corrected is switched according to the time elapsed after the generation of the tone signal is instructed, so that a reliable sound is produced and the sound is produced. The expression range of can be expanded.

In particular, the tone color of the wind instrument can be played by the electronic musical instrument by using the tone generation region characteristic diagram of the wind algorithm. In this case, the expressiveness is richer than playing with only the keyboard. It is easier to play than an actual wind instrument, and the performer's breathing is easy because the sound is reliably generated by operating the operating elements. With wind instruments, the sound is adjusted by the degree of bite of the reed and the overwhelming breath when playing, but the sound can be easily adjusted because the operator only moves the hand. Since the hand can be freely moved on the operator, a wide dynamic range can be secured. You can easily control the volume and sound quality. In addition, it is possible to sensuously match the motion for pronunciation with the sound. Furthermore, the duration of the sound can be lengthened.

[Brief description of drawings]

FIG. 1 is a basic configuration diagram of an electronic musical instrument control mechanism according to the present invention, FIG. 2 is a sounding region characteristic diagram of a rub string algorithm, FIG. 3 is an explanatory diagram of a method of switching between rising process and continuous process, and FIG. FIG. 5 is a diagram showing the relationship between bow pressure and bow speed of a stringed instrument, FIG. 5 is a basic configuration diagram of an electronic musical instrument according to the present invention, FIG. 6 is a flow chart of a program control main routine according to the method of the present invention, and FIG. 7 is a mode. Fig. 8 is a flow chart at the time of switching, Fig. 8 is a top view of the slide volume type performance operator, and Figs. 9 (a) and 9 (b) are side views and a top view of essential parts of the performance operator of Fig. 8, Fig. 10. Is a flow chart when the key is on, FIG. 11 is a flow chart when the key is off, FIG. 12 is an explanatory diagram when selecting a tone color, FIG. 13 is an explanatory diagram of a channel register table, FIG. 14 is a flowchart of a timer interrupt routine, Figure 15 is a flow chart of the sound source control routine, Is an explanatory view of the read management data table, Fig. 17 is a block diagram of the tone generator circuit of a stringed instrument, Fig. 18 is a sounding region characteristic diagram of a wind algorithm, and Fig. 19 is a basic configuration of a tone generator control mechanism of an electronic wind instrument using a 3D tablet. Fig. 20, Fig. 20 is a flow chart of the process switching operation at the time of rising and sustaining the sound of the electronic wind instrument, Fig. 21 is a block configuration diagram of the sound source control mechanism of the electronic wind instrument, Fig. 22 is a basic configuration diagram of the electronic wind instrument, FIG. 23 is an explanatory view of the main flow of the sound source control program, FIG. 24 is an operation explanatory view at the time of key-on, FIG. 25 is a flow chart at the time of assigning a breath pressure related device, FIG. 26 is a flow chart of the lateral shaking effect, FIG. 27 is a flow chart of the panel switch processing, FIG. 28 is a flow chart of the first example of the interrupt routine, FIG. 29 is a flow chart of the second example of the interrupt routine, and FIGS. 30 and 31 are the embouchure. And breath pressure 32 is a flow chart of the delay length parameter processing routine, FIG. 33 is a flow chart of the loop gain processing routine, FIG. 34 is a flow chart of the arithmetic processing routine, and FIG. 35 is a breath pressure. The flow chart of the first example of the correction routine, FIG. 36 is the flow chart of the second example of the breath pressure correction routine, FIG. 37 is the block diagram of the sound source of the wind instrument algorithm, and FIG. 38 is the breath pressure of the wind instrument algorithm. The relationship graph of Embouchure, FIGS. 39 (a) and 39 (b) are graphs of the relationship between the time at which the sound rises and the volume when the pressure is high and low, respectively. In FIG. 40, (a) and (b) are respectively. FIG. 41 is a control explanatory view of the lateral shaking effect in the wind instrument algorithm, and a graph of the relationship between time and volume when the sound disappears when the pressure is large and when the pressure is small. 1,15,27,74: Performance controller, 4: Correction circuit, 5,13,78: Keyboard, 6,23: Sound source, 18: CPU, 76: Conversion program.

Claims (3)

[Claims] 1. A sound source means for generating a tone signal based on a plurality of types of parameters generated by a parameter generating means, the parameters generated by the parameter generating means being defined by the plurality of types of parameters. A method of controlling an electronic musical instrument sound source, which comprises correcting the soundable area of the sound source means and inputting the corrected parameters. 2. An electronic musical instrument having parameter generating means for generating a plurality of types of parameters and sound source means for generating a tone signal based on a plurality of types of parameters generated by the parameter generating means. The parameters generated from
An electronic musical instrument, comprising: a correction unit that corrects a parameter of a soundable region of the sound source unit defined by the plurality of types of parameters and inputs the corrected parameter to the sound source unit. 3. The electronic musical instrument according to claim 2, further comprising a sound production instructing means for instructing the generation of a musical tone signal, wherein the correcting means comprises: (a) generation of a musical tone signal instructed by the sound producing instructing means. Within a predetermined time from, the parameter generated from the parameter generating means is corrected to the parameter of the sounding area of the soundable area of the sound source means, and (b) the sounding instructing means instructs generation of a tone signal. An electronic musical instrument characterized in that, after a predetermined time has passed, a parameter generated by the parameter generating means is corrected to a parameter of a continuous area other than the sounding area in the soundable area of the sound source means.