JPH10228281A - Electronic musical instrument - Google Patents

Abstract

PROBLEM TO BE SOLVED: To provide an electronic musical instrument which can simulate even the longitudinal vibration of a string with simple constitution in addition to the lateral vibration. SOLUTION: This instrument is provided with 1st and 2nd closed-loop circuits 30 and 30H having delay circuits 31 and 35, and 31H and 35H connected in closed loops, an exciting circuit 50 which generates an exciting signal, supply means (adders 32 and 36, and 23H and 26H) which supply the exciting signal to the 1st closed-loop circuit 30 and also raises the exciting signal to power, and supplies the power-raised exciting signal to the 2nd closed-loop circuit 30H, and an output means (adder 85) which puts the signal circulated in the 1st closed-loop circuit 30 and the signal circulated in the 2nd closed-loop circuit 30H together and outputs the resulting signal as a musical sound signal.

Description

DETAILED DESCRIPTION OF THE INVENTION

[0001]

BACKGROUND OF THE INVENTION 1. Field of the Invention The present invention relates to an electronic musical instrument, and more particularly to a keyboard electronic musical instrument such as a piano.

[0002]

2. Description of the Related Art A keyboard is the most common operation means of an electronic musical instrument, and is an excellent operation means that is easy for a player to handle and at the same time easy to express emotions. 2. Description of the Related Art Conventionally, there has been known a keyboard electronic musical instrument which generates a detection signal corresponding to a touch when a key on a keyboard is pressed, and controls the intensity of a musical tone based on the detection signal.

[0003] Also, there have been proposed various tone synthesizers having an electric model or a software model simulating a sounding mechanism of a natural musical instrument and operating these models. By inputting a detection signal generated by pressing a keyboard to this kind of musical sound synthesizer, a keyboard electronic musical instrument capable of performing a more rich performance can be realized.

[0004] In order to enhance the expressive power of a musical instrument, it is necessary that a touch when a player presses a key is faithfully reflected in a musical tone. FIG. 37 shows a schematic configuration relating to key press detection in a conventional keyboard electronic musical instrument for one key. As shown in this figure, two switches SW1 and SW2 for detecting that a key has been pressed are attached to each key KEY. Here, the switch SW1 is turned on when the key KEY is pressed down to the first depth, and the switch SW2 is turned on when the key KEY is pressed down to the second depth which is deeper than the first depth. . By pressing the key KEY,
When the switches SW1 and SW2 are sequentially switched to the ON state before and after the time, the time difference of the switching timing of each switch to the ON state is counted by the time difference detection circuit DET, and the key KEY is pressed based on the count result. A key speed signal corresponding to the key speed is generated. And C
The PU (central processing unit) 1 controls the sound intensity of the sound source TG based on the key speed signal.

[0005]

In a natural instrument such as a piano, vibrations propagating through a string include lateral vibration having an amplitude in a direction perpendicular to the string and amplitude in a horizontal direction (axial direction) of the string. And a longitudinal vibration having Here, the longitudinal vibration is a compression wave generated by the string extending in the vertical direction when the hammer collides with the string, and propagates at a speed several tens times faster than the transverse vibration. In fact, if you play the piano hard,
You can hear a characteristic sound with a pitch about ten times the fundamental tone. This is what is called a string sound caused by the longitudinal vibration of the strings. The bending sound is hardly heard at the time of a weak touch, but rapidly increases at the time of a strong touch (the strength of the bending sound is the square of the touch or the square of the lateral amplitude). However, in a keyboard electronic musical instrument using a conventional electric model or software model, generally, only the lateral vibration of a string is simulated. This is because simulating not only lateral vibration but also longitudinal vibration leads to an enormous amount of calculation and is not realistic. For this reason, in a general electronic musical instrument, there has been a problem that it is not possible to simulate the stringing sound caused by the longitudinal vibration of the strings as described above.

The present invention has been made in view of the above circumstances, and has as its object to provide an electronic musical instrument that can simulate not only horizontal vibration but also vertical vibration with a simple configuration.

[0007]

SUMMARY OF THE INVENTION In order to solve the above-mentioned problems, the invention according to claim 1 comprises first and second closed loop means in which at least delay means are connected in a closed loop, and an excitation signal generating means for generating an excitation signal. Means for supplying the excitation signal to the first closed loop means, powering the excitation signal, and supplying the raised excitation signal to the second closed loop means, and the first closed loop Output means for synthesizing a signal circulating through the means and a signal circulating through the second closed-loop means and outputting as a tone signal. Further, the invention according to claim 2 is the first invention wherein at least the delay means is connected in a closed loop.
And second closed loop means, excitation signal generating means for generating an excitation signal, first supply means for supplying the excitation signal to the first and second closed loop means, and circulating through the first closed loop means A second supply means for supplying a signal to be raised to the second power to the second closed loop means, a signal circulating in the first closed loop means and a signal circulating in the second closed loop means, and a tone signal And output means for outputting as

[0008]

According to the first aspect of the present invention, the excitation signal is supplied to the first closed loop means by the supply means, the excitation signal is raised to the power, and the raised excitation signal is supplied to the second closed loop means. Is done. According to the second aspect of the present invention, the excitation signal is supplied to the first and second closed loop means by the first supply means,
The signal circulating through the first closed loop means is raised to the power by the second supply means and supplied to the second closed loop means.
Here, the signal corresponding to the horizontal vibration of the string circulates in the first closed loop means, and the signal corresponding to the vertical vibration of the string circulates in the second closed loop means. Then, the signal circulating through the first closed-loop means and the signal circulating through the second closed-loop means are synthesized by the output means and output as a tone signal.

[0009]

Embodiments of the present invention will be described below with reference to the drawings. FIG. 1 is a block diagram showing a schematic configuration of a keyboard electronic musical instrument according to one embodiment of the present invention. The CPU 1 reads control parameters from the parameter memory 2 as necessary, and supplies control information to each section of the electronic musical instrument. KEY
1 to KEY2 are keys provided on the keyboard, respectively, and acceleration pickups 24 are attached to the respective tips. TG1 to TGn are key press detection / tone synthesizers that form a tone in response to a key press operation on each of the keys KEY1 to KEYn. These key press detection / tone synthesis sections TG1 to TG
n is half-wave rectifier circuits 11 and 12, integrating circuit 13, A
/ D (analog / digital) converter 14, tone synthesizer 1
5 The half-wave rectifier circuit 11 receives the acceleration detection signal output from the acceleration pickup 24, selects only the positive component thereof, and outputs the selected positive component. This positive component is integrated by the integration circuit 13 and the integration result is output to the A / D converter 1.
4 is converted into a digital signal. A / D converter 1
The output digital signal of No. 4 is supplied to the musical tone synthesizer 15 as a hammer speed signal HV corresponding to the speed of the hammer in the piano. In the tone synthesizer 15, a tone signal is synthesized based on the hammer speed signal HV. Each tone signal output from each tone synthesis unit 15 of the key press detection / tone synthesis unit TG1 to TGn is added by adders A2 to An.
The signal is input to one input terminal of the adder 5.

On the other hand, key press detection / tone synthesis sections TG1 to TG
n each half-wave rectifier circuit 12 selects only the negative component in the acceleration detection signal, and when the key hits the pedestal (the so-called "dumb", which is the tree under the keyboard) It is output as a signal corresponding to the acceleration received by the keyboard (also referred to as the acceleration received by the pedestal). Each half-wave rectifier circuit 1
2 are added by adders B2 to Bn, and A /
The signal is converted into a digital signal by the D converter 3. The output signal of the A / D converter 3 is input to a filter 4 which approximates a piano frame or the like. Then, a signal corresponding to so-called mechanical noise is output from the filter 4,
The signal is input to the other input terminal of the adder 5. Then, the adder 5 sums the sum of the outputs of the respective tone synthesis units 15 and the filter 4.
Is added to the output. The result of the addition passes through a resonance circuit 6 simulating a piano frame, a soundboard, and the like, and is then converted into an analog signal by a D / A (digital / analog) converter 7 and supplied to a speaker SP, and a musical tone is produced. Pronounced.

The detailed configuration and operation of each part of the keyboard electronic musical instrument will be described below in order. [Circuit Configuration for Key Press Detection] FIG. 2 shows the mounting structure of keys KEYj (j = 1 to n) in this keyboard electronic musical instrument. As shown in the figure, the key KEYj is attached to the keyboard so that it can rotate around the fulcrum 22. Key KEYj
The acceleration pickup 24 is attached to the tip of the side on which is pressed by the player. Also, the key KE
A spring 23 is attached to the tip of Yj on the side opposite to the side pressed by the player, and this spring 23 gives a moment for lifting the tip on the side pressed by the player upward. Note that a weight may be attached instead of the spring 23. Acceleration pickup 24
The attachment position may be any position as long as it is at a certain distance from the key fulcrum 22. By performing weighting according to the distance from the fulcrum 22, an equivalent acceleration detection signal ACC can be obtained for each attachment position. However, in order to accurately detect the acceleration, the acceleration pickup 24
Is preferably attached to the tip as much as possible. A switch 25 is provided below the tip of the key KEYj. The switch 25 is turned on when the key KEYj is depressed by a predetermined amount, and a key switch signal KON / KOFF having a signal value corresponding to the on / off state of the switch 25 is generated. The key switch signal KON / KOFF is an instruction to take in the acceleration detection signal ACC in the integrator 13 and start the integration operation.
It is used as a trigger of the tone synthesis process in the tone synthesis unit 15. Therefore, the key switch signal KON / KOFF
Must be output without delay with respect to the key depression operation, and for that purpose, it is better to mount the switch 25 at a position as shallow as possible.

FIG. 3 shows a detailed configuration example of a circuit for processing the acceleration detection signal ACC output from the acceleration pickup 24. FIG. 2 shows a subtractor 16, a switch 17, and a differentiator 18, which are not shown in FIG. When the acceleration detection signal ACC passes through the half-wave rectifier circuit 11, its positive component POS (key KE
(Corresponding to the case where Yj moves downward) only, and is input to the integrator 13. When the key KEYj is pressed, the switch 25 is turned on and the key switch signal KON / KOFF rises. The rise of the key switch signal KON / KOFF is detected by the differentiator 18, a trigger signal is sent from the differentiator 18 to the integrator 13, and the integrator 13 starts the integration operation. Then, the integrated waveform KW of the integrator 13 is sequentially converted into a digital signal by the A / D converter 14, and the hammer speed signal HV
Is supplied to the tone synthesizer 15. When the pressed key KEYj is released, the switch 25 is turned off and the key switch signal KON / KOFF falls.
The fall of the key switch signal KON / KOFF is detected by the differentiator 18, the integrator 18 is reset by the differentiator 18, and the apparatus enters a state of waiting for the next key pressing operation. FIG. 4 shows an acceleration detection signal ACC output from the acceleration pickup 24 and an output waveform KW of the integrator 13.

On the other hand, as the acceleration signal ACC passes through the half-wave rectifier circuit 12, only the negative component NEG is extracted. Then, the offset OFFSET corresponding to the moment by the spring (or the weight) is subtracted from the negative component NEG by the subtractor 16, and the switch 17, the adder B2
ＢBn (FIG. 1) and the A / D converter 3, and is input to a filter 4 (FIG. 1) for generating mechanical noise. Key KEYj
When the key is released and the key switch signal KON / KOFF falls, the switch 17 is turned off, and the supply of the signal to the filter 4, that is, the generation of mechanical noise ends.

In FIGS. 1 and 3, the A / D converter 14 does not need to be very accurate. Therefore,
Instead of being provided for each key as shown in FIG.
Integrated waveform KW corresponding to each key using A / D converters
May be A / D converted in a time-division manner. For example, if the keyboard has 88 keys, the sampling frequency of the A / D converter 14 is set to 88.
kHz, and in each time slot obtained by dividing the sampling period into 88, A / A of the integrated waveform KW corresponding to each key.
Perform D conversion. In this way, for each key,
A hammer speed signal HM having a sampling frequency of 1 kHz is obtained.

According to the configuration described above, a hammer speed signal HM that is faithful to a touch when a key is pressed can be obtained. FIGS. 5A to 5D show the output POS of the half-wave rectifier circuit 11 and the integral waveform KW of the integrator 13 at the time of a typical touch. FIG. 5A shows each waveform when the player hits quickly and strongly. In this case, a large acceleration acts on the key KEYj from the time the button is pressed until the collision with the pedestal. Accordingly, the integral waveform KW of the integrator 13 rises greatly, and the large hammer speed signal HV is generated by the tone synthesis unit 1.
5 is supplied. FIG. 5B shows waveforms when the player hits quickly and weakly. In this case, the acceleration signal ACC rises to a large signal value, but the impulse given to the key KEYj is small, and the acceleration signal AC after the rise is increased.
C falls to a small signal value. Therefore, the integral waveform K
The final value of W becomes smaller and the hammer speed signal HV becomes smaller. FIG. 5 (c) shows the waveforms when the player hits slowly and strongly. In this case, the acceleration signal ACC at the rise is small, but the impulse given to the key KEYj is large, so that the integral waveform KW rises to a large value and the hammer speed signal HV increases. FIG. 5D shows each waveform when the player hits the ball slowly and weakly. In this case, since the acceleration signal ACC has a small value throughout, the hammer speed signal H
V becomes smaller. Thus, the hammer speed intended by the player can be input from the keyboard.

[Tone Synthesizing Unit] The tone synthesizing unit 15 is the most characteristic component of the present invention. First, the background art will be described, and then the configuration according to the present invention will be described in detail. FIG. 6 is a block diagram showing an example of a basic configuration of the musical sound synthesizer 15 as the background art of the present invention. In this figure, 30 is a loop circuit simulating the behavior of a string in a piano, and 50 is an excitation circuit simulating the behavior of a hammer. The tone synthesizer 15 includes a multiplier 43 that mediates a signal injected from the excitation circuit 50 into the loop circuit 30, and an adder 41 and a multiplier 42 that mediate a signal that is fed back from the loop circuit 30 to the excitation circuit 50.
And

The loop circuit 30 includes a delay circuit 31, an adder 3
2, a filter 33, a phase inversion circuit 34, a delay circuit 35,
The adder 36, the filter 37 and the phase inversion circuit 38 are connected in a closed loop. The outputs of the delay circuits 31 and 35 are taken out, added by an adder 41, and fed back to an excitation circuit 50 via a multiplier 42. Loop circuit 3
The extraction point of each of these outputs at 0 corresponds to the striking point P where the hammer HM strikes the string STR in FIG. That is, in the loop circuit 30, the adder 3
The delay time of the path from the input of the string 6 to the output of the delay circuit 31 is represented by the string striking point P in the string STR and one fixed end T
The length of the path from the input of the adder 33 to the output of the delay circuit 35 is equal to the delay time of the string striking. It corresponds to the delay time required for the vibration to reciprocate between the point P and the other fixed end T2 (length L2). The phase inversion circuits 38 and 34 are provided to simulate the phenomenon that the phase of the vibration wave is inverted at the fixed ends T1 and T2. Filters 37 and 3
3 is provided to simulate the acoustic loss when the vibration of the string STR is directly radiated into the air and when the vibration of the string STR propagates to the sound board of the piano via the fixed ends T1 and T2. It was done. Usually, this type of acoustic loss increases as the frequency increases, so that low-pass filters are used as the filters 37 and 33.

Next, the excitation circuit 50 will be described. The hammer speed signal HV output from the A / D converter 14 (see FIG. 1) is input to one input terminal of the adder 55. The other input terminal of the adder 55 receives the integrated value of the integrator 56. This integral value corresponds to a speed change generated in the hammer HM due to the interaction between the hammer HM and the string STR. The details of the calculation of the speed change will be described later. From the adder 55, a signal obtained by correcting the hammer speed signal HV by the change in the speed, that is, a signal corresponding to the current speed of the hammer HM is obtained. Then, the output signal of the adder 55 is integrated by the integrator 57, and a hammer displacement signal HD corresponding to the displacement of the hammer HM is output.

On the other hand, the adder 52 comprises multipliers 42 and 53
Are output. Here, the output signal of the multiplier 42 corresponds to the speed of the string STR at the string striking point P in FIG. 12, and the output signal of the multiplier 53 corresponds to the speed correction provided to the string STR by the hammer HM. Accordingly, the string STR at the current time of the string STR from the adder 52 is calculated.
The signal SV corresponding to the speed is output. When the signal SV is integrated by the integrator 54, the string S
A string displacement signal SV corresponding to the displacement of TR is obtained. Then, the string displacement signal SD is subtracted from the hammer displacement signal HD by the subtractor 58, and a relative displacement signal SHD corresponding to the bite amount of the string STR with respect to the hammer HM is obtained.

The relative displacement signal SHD is input to a multiplier 61, a nonlinear circuit 62, and a differentiator 64. Nonlinear circuit 6
2 is realized by a ROM, for example, and has a nonlinear input / output response characteristic as illustrated in FIG. As shown in this figure, the output of the non-linear circuit 62 increases with an increase in the input signal value, but the slope decreases as the input signal value increases. The multiplier 61 outputs the relative displacement signal SH
D is multiplied by a multiplication coefficient S according to the elasticity of the hammer HM and output. Then, the multiplier 63 multiplies the output of the multiplier 61 by the output signal of the nonlinear circuit 62.
As a result, the hammer HM may be deformed due to the elastic characteristics of the hammer HM.
And a signal corresponding to the repulsion generated between the strings STR is output from the multiplier 63. The output of the multiplier 63 increases as the relative displacement signal SHD increases. However, when the relative displacement signal SHD increases, the output of the nonlinear circuit 62 saturates, so that the output of the multiplier 63 also saturates. Thus, an operation faithful to the behavior of the actual hammer HM caused by the elasticity is obtained. On the other hand, a signal obtained by differentiating the relative displacement signal SHD by the differentiator 64 is multiplied by a multiplier 65 by a multiplication coefficient R according to the viscosity of the hammer HM. The output signal of the multiplier 65 is applied to the multiplier 66.
And 67, the output of the nonlinear circuit 62 is multiplied twice. By performing the multiplication twice, the output signal of the multiplier 65 is effectively multiplied by the signal obtained by performing the nonlinear conversion shown in FIG. 8 on the relative displacement signal SHD. As a result, a signal corresponding to the repulsion generated between the hammer HM and the string STR due to the viscosity of the hammer HM is output from the multiplier 65. The signal value of the output signal of the multiplier 65 is
The larger the temporal change of the relative displacement signal SHD, the larger the value. Further, even if the temporal change rate of the relative displacement signal SHD is the same, the output signal value of the multiplier 67 increases as the relative displacement signal SHD increases, that is, as the string STR digs deeper into the hammer HM. It will be. In this way, an operation faithful to the behavior of the actual hammer HM caused by the viscosity can be obtained. The outputs of the multipliers 63 and 67 are added by the adder 68, and the hammer HM and the string ST
A signal F corresponding to the repulsive force between the signal R and the signal R is output from the adder 68.

The output signal F of the adder 68 is input to the multiplier 43 and is multiplied by a multiplication factor 1/2. As a result, FIG.
In 2, the velocity component of the vibration wave propagating to both sides of the striking point P of the string STR is output from the multiplier 43. The output signal of the multiplier 43 is fed back to the adders 32 and 36 of the loop circuit 30, while being multiplied by a predetermined multiplication coefficient FADM by the multiplier 53, and the string STR by the hammer HM.
Is output from the multiplier 53. The output signal F of the adder 68 is multiplied by a multiplier coefficient −1 / M (where M is the mass of the hammer HM) by the multiplier 69, and a signal HA corresponding to the acceleration acting on the hammer HM is output. . This signal HA is integrated by the integrator 56, and a signal corresponding to the above-mentioned speed change of the hammer HM is obtained.

The operation of the tone synthesizer 15 will be described below. The key KEYj corresponding to the tone synthesis unit 15 is depressed, and the corresponding key switch signal KON / KOFF
Rises, the initial value 0 is preset in the integrators 56 and 57, and the simulation is started from a state where the hammer HM collides with the string STR. Then, the hammer speed signal HV corresponding to the key pressing operation is output from the A / D converter 14.
Output from This hammer speed signal HV is added to the adder 5
5, the signal is input to the integrator 57, is integrated, and the hammer displacement signal HD is output. And the hammer displacement signal HD
Is input to a subtractor 58, and a relative displacement signal SHD is output. Then, as described above, the relative displacement signal SHD
Is generated, and based on the signal F, a signal HA and a hammer HM corresponding to the acceleration of the hammer HM are generated.
Signals corresponding to the speed change of the hammer H
The signal corresponding to the current speed of M (the output of the adder 55) is corrected. On the other hand, the signal F is fed back to the loop circuit 30 via the multiplier 43, and is further added to the multiplier 53,
The signal is input to the integrator 54 via the adder 52 and the adder 52 and is integrated. As a result, the integrated value of the integrator 54, that is, the signal corresponding to the displacement of the string STR is corrected. The signal input from the excitation circuit 50 to the adder 36 in the loop circuit 30 is extracted again from the loop circuit 30 via the filter 37, the phase inversion circuit 38, and the delay circuit 31.
On the other hand, the signal input to the adder 32 is
It is again taken out of the loop circuit 30 via the phase inversion circuit 34 and the delay circuit 35, integrated by the adder 41, and fed back to the excitation circuit 50 via the multiplier 42. As a result, the signal SV corresponding to the speed of the string STR is corrected, and the signal SD corresponding to the displacement of the string STR is corrected. Thereafter, similarly, the excitation circuit 50 and the loop circuit 30
Thus, a simulation of the interaction between the hammer HM and the string STR and a simulation of the propagation of vibration in the string STR are performed. Then, a signal corresponding to the vibration velocity component of the string STR is extracted from an arbitrary node in the loop circuit 30 and output as a tone signal.

In the above description, the simulation is started from the moment when the hammer HM collides with the string STR. However, in the initial state, the hammer HM is separated from the string SP, and the hammer HM moves toward the string STR by pressing a key. If the simulation is performed including the state of starting to move, the actual piano sound can be reproduced more faithfully. In this case, the key switch signal KON / KO
When the FF rises, the configuration is changed so that an initial displacement indicating the distance between the hammer HM and the string STR is preset in the integrator 57 of the tone synthesis unit 15 (FIG. 6). Along with this change, the configuration of the circuit related to key press detection shown in FIG. 3 is changed to that shown in FIG. That is, a subtractor 19 that subtracts the signal g corresponding to the gravitational acceleration from the output of the half-wave rectifier circuit 11 and supplies the signal to the integrator 13 is added to the configuration of FIG. Thus, it is possible to simulate the hammer HM returning without reaching the string STR when the touch is considerably weak. In addition, it is possible to reproduce a state in which the hammer HM barely collides with the string STR during pianissimo.

[Another Configuration Example of the Tone Synthesizing Unit] The filters 33 and 37 of the tone synthesizing unit shown in FIG. 6 are constituted by an FIR (finite impulse response) filter, an IIR (infinite impulse response) filter, an all-pass filter, and the like. Is possible. However, when low-order filters are used as these filters, the degree of freedom in the time direction decreases. Therefore, in this case, the frequency at which the vibration propagates back and forth through the string STR is adjusted by the delay time of the delay circuits 31 and 35. For this reason, the state of the vibration of the string STR can be approximated to some extent on the frequency axis, but the phase characteristic of the vibration of the string STR can hardly be approximated. On the other hand, a real piano string S
The TR has elasticity, and has a dispersive property that the higher the frequency of the vibration, the faster the propagation. FIG. 10 shows an example of a string impulse response. As shown in the figure, in an actual piano, a high-frequency precursor wave appears during a period τ ₀ from when a string is struck to when a main pulse appears on the string. For this reason, the vibration of the strings is incongruent, and this incongruity is one factor for generating a piano-like sound. Focusing on the interaction between the hammer HM and the string STR, the waves reflected at the fixed ends T1 and T2 and field-backed to the stringing point P while the hammer HM is in contact with the string STR are reflected on the hammer HM. It is necessary to accurately reproduce the applied force. Since the hammer HM has viscosity and elasticity, the hammer HM and the string STR move while being in contact for a very short time. During this time, the hammer H
If the force exerted on M can be faithfully simulated, the interaction of the hammer HM with the string STR is accurately described, and a realistic piano sound is realized.

In the following, in consideration of the above, various models of a musical sound synthesizer capable of synthesizing a musical sound faithful to an actual piano sound will be described. First, a model <1> in which the impulse response including the phase characteristic of the string STR is accurately approximated by a high-order FIR filter will be described. Next, a model <2> in which the phase characteristics of the strings are approximated by an all-pass filter
Will be described. Further, in the model <1>, there is an inconvenience that the decay is shortened due to an error generated when obtaining the coefficients of the FIR filter. The inconvenience is improved by combining the models <1> and <2> with the model <3>. State. In addition, as an improved example of the musical sound synthesizing unit, a model <4> according to the present invention that enables synthesis of a hinari sound, a model <5> that simulates pronunciation by plural strings, and a model <6 corresponding to continuous pronunciation. >
Will be described sequentially. Further, a model <7> that takes into account not only the hammering of the strings but also the muting of the dampers will be described.

Model <1>: Musical Sound Synthesizing Unit Using Higher Order FIR Filter FIG. 11 is a block diagram showing a configuration example of a musical sound synthesizing unit according to this model <1>. This tone synthesizer uses a high-order FIR filter as the filters 33 and 37 in FIG. These FIR filters realize a transfer function of a piano string represented by the following equation (1).

(Equation 1) Here, τ ₀ indicates that the main pulse has changed to the string S after the string was struck.
Delay time before appearing in TR, γ is a coefficient related to the anharmonicity of the vibration of the string STR, a is a coefficient corresponding to the air friction acting on the string STR, b is the time when the vibration of the string STR radiates into the air Is a coefficient corresponding to the radiation characteristic of The transfer function is described in The University of ElectroCommunicati
ons “Piano Tone Synthesis Using Digital Filters B
This is explained in "Y Computer Simulation" (by Nakamura) and in the Proceedings of the Acoustical Society of Japan, "Application of Digital Filter Method to Vibration of Interacting Piano Strings" (by Nakamura). The impulse response of the piano string is obtained by performing an inverse Fourier transform on the transfer function of Equation (1). By sampling this impulse response, multiplication coefficients a0 to an and b0 to bn of each FIR filter are obtained. The transfer function of the string is mainly determined by the pitch and the degree of inharmonicity, and thus the FI
The number of stages and coefficients of the R filter are determined. Excitation circuit 5
The signal F from 0 (corresponding to the force when the hammer HM collides with the string STR) has a sharp rise and therefore has a high frequency component. According to the configuration of FIG. 11, since a signal containing a large amount of the high-frequency component passes through the high-order FIR filters 33 and 37, a considerable precursor wave appears in the signal waveform circulating through the loop circuit 30, and this is the loop circuit. It is taken out from 30 and fed back to the excitation circuit 50. As a result, the tone signal waveform extracted from the tone synthesizer becomes very close to the actual piano sound.

When the processing of the tone synthesizer having the configuration shown in FIG. 11 is realized by software, if the entire delay time of the loop circuit 30 is realized by FIR filter operation, a considerable amount of operation is required. Therefore, if the impulse response is considerably attenuated in order to reduce the amount of calculation, the FIR filter calculation may be substituted by a plurality of stages of delay calculation. Further, it is possible to improve the degree of freedom of the timbre by combining delay, low-order filters, and the like. As a low-order filter, an FIR low-pass filter, an FIR high-pass filter, an IIR low-pass filter, an all-pass filter, or the like can be used.

Model <2>: Musical Sound Synthesizing Unit Using All-Pass Filter Before describing the model <2>, a general first-order all-pass filter will be described. Figures 13 and 1
4 shows a configuration example of a primary all-pass filter. Here, the all-pass filter is a filter in which the gain is always exactly 1 regardless of the input signal frequency, and only the phase delay depends on the frequency. The amount of phase delay of the all-pass filter is determined by the multiplication coefficient C of the multiplier in the filter. An example of the transfer function of a first-order all-pass filter is given by Equation 1.
Shown in Equation 3 shows a phase characteristic equation based on the transfer function of Equation 1. Further, FIG. 15 shows the frequency characteristic of the delay amount (phase delay) when the coefficient C is changed between [−1 <C <1] in the phase characteristic equation of Expression 3.

(Equation 2)

(Equation 3) In the above, T is the delay time of the one-sample delay circuit. As shown in FIG. 15, as C approaches -1, the amount of delay at low frequencies increases sharply. However, the delay amount of the high frequency is almost only one sample period.

FIG. 16 is a plot of the phase characteristics of an actual piano string. As shown in this figure, in a piano string, the higher the frequency, the smaller the delay amount and the faster the string propagates. Therefore, if the loop circuit 30 is configured by connecting all-pass filters in multiple stages so that a phase delay corresponding to the delay amount (the number of samples) corresponding to the basic pitch of the musical tone can be realized, the phase characteristics of the piano strings can be simulated. Can be. However, since the delay amount corresponding to the basic pitch is obtained by dividing the period of the musical tone waveform of the target pitch by the sampling period, the delay amount is usually several hundreds of samples. The amount of delay is at most a few samples. Therefore, a method of giving a phase delay to frequency components of several hundred kilohertz or less by an all-pass filter of about several tens of stages and giving a phase delay to a signal of the remaining frequency components by a delay circuit is considered.

There is also a method using a higher-order all-pass filter instead of the first-order all-pass filter. FIG.
7 is a configuration example of a second-order all-pass filter, and FIG. 18 is a general form of a multi-order all-pass filter. Also, FIG.
This is an all-pass filter with a Lattice-type configuration.
0 indicates the configuration of the m-th element. In addition,
Another configuration example of the first-order and second-order all-pass filters is described in "Basics of Digital Signal Processing (supervised by Shigeo Tsujii: The Institute of Electronics, Information and Communication Engineers)" and the like. Since the multi-order all-pass filter can set the peak frequency of the delay amount and its steepness, it is possible to perform a fairly accurate simulation of the striking operation of the piano by combining it with the first-order one.

Hereinafter, an example of the configuration of a tone synthesis unit based on the present model <2> will be described. This model <2> has the same structure as that of FIG.
Is replaced by a cascade connection of a multi-stage all-pass filter and a delay circuit. The signal F from the excitation circuit 50 (corresponding to the force applied to the string STR from the hammer HM) has a sharp rising and high frequency component. When the signal F passes through the all-pass filter, a difference is generated in each phase delay of each frequency component, and the higher the frequency component, the faster it circulates in the loop circuit 30 in time. For this reason, in the musical tone waveform extracted in a loop of the loop circuit 30, a considerable precursor wave appears before the main pulse. This causes fine wrinkles (high-frequency components) in the signal waveform circulating in the loop circuit 30, which are fed back to the excitation circuit 50. As a result, an operation in which a force due to a specific high frequency component is fed back from the string STR to the hammer HM is simulated. As a result of such an operation, a faithful musical tone is synthesized with the actual piano sound.

Model <3>: tone synthesizer with improved decay As shown in model <1> above, the transfer characteristic of string
When implemented with filters, these filters are preferably all-pass (all-pass). Also,
Ideally, the operation corresponding to the attenuation of the vibration of the string STR in the tone synthesizer is realized by setting the multiplication coefficient of the phase inversion circuits 34 and 38 corresponding to the pieces to a value slightly larger than -1. . However, since each multiplication coefficient of the FIR filter obtained from the impulse response of the string STR includes an error, it is difficult to realize all-pass characteristics, and an error occurs in the amplitude response depending on the frequency. In this case, the model <1> corresponding to the treble range is F
Since the order of the IR filter is low, the error of the amplitude response increases as shown in FIG. 22 and fluctuates in a large cycle. Conversely, the order of the FIR filter of the model <1> corresponding to the bass range becomes higher, so that the period of the error of the amplitude response becomes smaller as shown in FIG. However, in the model <1> (the configuration of FIG. 11) that divides the string into two parts at the striking position and simulates the operation of each divided part, the impulse response of one string is divided into two FIRs.
It is realized by a filter. For this reason, the order of one FIR filter becomes low, and eventually, the error in the amplitude response becomes large. Therefore, the decay becomes short at a frequency where the amplitude response is large and less than 1, and the envelope of the overtone is different from the actual piano sound.

Hereinafter, means for solving this problem will be described. The reason why the simulation is performed by dividing the string STR into two in the model <1> is to accurately simulate the interaction during the period in which the hammer HM and the string STR collide. Therefore, if the hammer HM separates from the string STR and the interaction between the two ends, there is no need to divide the simulation into two, and the behavior of one string can be simulated by one filter. Based on this concept, the structure of the model <1> is modified as described below to separate a part that simulates the interaction between the string STR and the hammer HM from a part that simulates the propagation of vibration in the string STR. Do.

FIG. 23 is a block diagram showing a configuration completely equivalent to the model <1> (FIG. 11). FIG. 23 and FIG.
In the FI corresponding to the part corresponding to the length L1 of the string STR
Although the positional relationship between the R filter 37 and the phase inverting circuit 38 and the positional relationship between the FIR filter 33 and the phase inverting circuit 38 corresponding to the portion corresponding to the length L2 are reversed, the same operation is performed in both. Done. First, in the configuration shown in FIG. 23, the adder 32 is a three-input adder 32a.
And the adder 36 is omitted. And F
FIR filter 37a and phase inverting circuit 38a having exactly the same configuration as IR filter 37 and phase inverting circuit 38
Is added, and the output of the multiplier 43 is input to the FIR filter 37a. The output of the multiplier 43, the output of the FIR filter 37, and the phase
The output of a is added by the adder 32a and supplied to the phase inversion circuit 34. With this change, the configuration shown in FIG. 24 is obtained. 23 and 24, the sum of the output signal of multiplier 43 and the signal obtained by passing this output signal through a FIR filter and a phase inversion circuit corresponding to length L1 is given by The signal is input to the phase inversion circuit 34. Therefore, in the configuration of FIG. 24, an operation completely equivalent to the operation obtained in the configuration of FIG. 23 is obtained. Next, in the configuration of FIG. 24, the supply of the output of the FIR filter 33 to the adder 41 is eliminated, and a series circuit including the phase inversion circuit 34a and the FIR filter 33a having the same configuration as the phase inversion circuit 34 and the FIR filter 33 is provided. Adder 32a
, So that the output of the FIR filter 33a is supplied to the adder 41. With this change, the configuration shown in FIG. 25 is obtained. 24 and 25, the output signal of FIR filter 37 and the signal obtained by the output signal of adder 32a passing through the phase inversion circuit and the FIR filter corresponding to length L2 are added. The signal is fed back to the excitation circuit 50 via the multiplier 42. Therefore, according to the configuration of FIG.
An operation completely equivalent to the operation in each configuration of FIGS. 24 and 23 is obtained. Thus, the string STR and the hammer HM
All the elements related to the interaction with the STR can be collected outside the loop, and finally, a high-order FIR filter corresponding to one string STR can be provided in the loop. As a result, it is possible to reduce the error of the amplitude response.

By replacing the FIR filters 33 and 37 and the phase inverting circuits 34 and 38 in the configuration of FIG. 25 with multi-stage all-pass filters 300-1 to 300-n, the model <3 shown in FIG.
> Is obtained. According to this model <3>, the hammer HM
Interaction between the string STR and the high-order FIR filter 37a
And the behavior of the string STR is simulated by the all-pass filters 300-1 to 300-300 in the loop.
Simulated by -n. In this way, an all-pass filter is inserted in the loop circuit in which the amplitude characteristic is important, and an FIR filter is inserted in a portion relating to the interaction between the hammer HM and the string STR in which the phase response is important. It is possible to achieve both accurate reproduction of the vibration waveform and accurate reproduction of the interaction between the hammer HM and the string STR.

Model <4>: Tone synthesizer capable of synthesizing a beat sound (a configuration which is the most characteristic of the present invention) A vibration propagating through a string STR has a lateral amplitude having an amplitude in a direction perpendicular to the string. There are vibrations and longitudinal vibrations having a horizontal (axial) amplitude of the strings. Here, the longitudinal vibration is the hammer H
This is a compression wave generated by the string STR extending in the vertical direction when M collides with the string STR, and propagates at a speed several tens of times faster than the transverse vibration. In fact, when you play the piano strongly, you can hear a characteristic sound whose pitch is about ten times as high as the fundamental tone. This is called a string sound caused by longitudinal vibration of the strings. The bending sound is hardly heard at the time of a weak touch, but rapidly increases at the time of a strong touch (the strength of the bending sound is the square of the touch or the square of the lateral amplitude). Each of the models described so far simulates only lateral vibration, but this model <4> simulates longitudinal vibration in addition to lateral vibration, and contains more actual This synthesizes musical tones that are close to the piano sound. However, since it is not realistic to rigorously synthesize the tongue sound, the amount of calculation is enormous, and in this model <4>, as shown in FIG.
In addition to the loop circuit 30 corresponding to the actual string STR, a loop circuit 30H for synthesizing a beat sound is added to the configuration of FIG. A loop circuit 3 for synthesizing this sound
0H is a delay circuit 31H, an adder 32H, a filter 33
H, adder 39H, phase inversion circuit 34H, delay circuit 35
H, an adder 36H and a phase inverting circuit 38H are connected in a loop, and the delay circuits 31H and 3H
5H has a smaller number of delay stages than the delay circuits 31 and 35, and the filter 33H is a low-order filter. Multiplier 4
3 (corresponding to the force exerted on the string STR by the hammer HM) is injected into the loop circuit 30 while being squared by the multiplier 81, and the result is input by the multiplier 82 to the input gain g. ₁ is multiplied and the adder 3
Injected into loop circuit 30H from 2H and 36H.
Further, a signal corresponding to the horizontal vibration of the string STR is picked up as a signal at the input terminal of the phase inversion circuit 34 corresponding to the bridge T2, and this signal is squared by the multiplier 83. Is multiplied by the gain g ₂ ,
The signal is injected from the adder 39H into the loop circuit 30H. Then, a signal corresponding to the horizontal vibration propagating through the loop circuit 30 and a signal corresponding to the longitudinal vibration (finish sound) propagating through the loop circuit 30H are respectively extracted, added by the adder 85, and output as a musical tone signal. You. When the string STR is extended to a plurality of strings, it is better to prepare a loop circuit for string sound synthesis corresponding to each string.

Model <5>: Musical tone synthesizer simulating sound generation with multiple strings An actual piano corresponds to one pitch and has a plurality of strings, and the characteristics of each of these strings are slightly shifted. It is common to have. Therefore, it is preferable that the tone synthesis section also prepares circuits simulating these strings, and slightly shifts the parameters of the circuits. By doing so, the musical sound has a chorus feeling and swell, and the likeness of the actual piano sound increases. Also, by configuring so that signals are transmitted and received between circuits simulating each string, resonance between a plurality of strings can be realized. FIG. 28 shows an example of the configuration of a tone synthesis unit based on the present model <5> that simulates sound generation by a plurality of strings. In this figure, 91 and 92 are waveguides (bidirectional transmission circuits) simulating each of two strings, and 93 is a waveguide simulating a resonance system such as a piano frame or a soundboard. Here, the transmission characteristics of the waveguides 91 and 92 are slightly shifted. The wave guide is disclosed in
This is described in JP-A-3-40199. FIG.
, The outputs of the waveguides 91 to 93 are multiplied by coefficients α _{1 to} α ₃ by multipliers 94 to 96, respectively, and the multiplication results are added by an adder 97. Here, each of the coefficients α ₁ , α ₂ and α ₃ has the following relationship.

(Equation 4)

The adder 202 causes the adder 97
Is added to a signal obtained by inverting the output signal of the waveguide 91 by the phase inversion circuit 201, and the addition result is fed back to the waveguide 91. Further, the adder 204 adds the output signal of the adder 97 and the signal obtained by inverting the output signal of the waveguide 92 by the phase inversion circuit 203, and the addition result is fed back to the waveguide 92. Further, the output signal of the adder 97 and the signal obtained by inverting the output signal of the waveguide 93 by the phase inversion circuit 205 are added by the adder 206, and the addition result is fed back to the waveguide 93.
According to such a configuration, transmission and reception of signals between the respective waveguides are performed, and resonance in two strings, a soundboard, and the like is simulated. The example of the case of two strings has been described above, but it goes without saying that the present model can be applied to the case of three strings and four strings. Further, the present model may be extended to simulate the interaction between strings for 88 keys.

Model <6>: Musical Tone Synthesizing Unit Compatible with Continuous Tone This model <6> adds a means for controlling tone generation to the configuration shown in FIG. 6 in response to continuous key press operation. It is realized by doing. That is, in the present model <6>, a. When the tip of the hammer HM separates from the string STR and returns to the position in the stationary state; b. When a key release is detected; c. When the next key depression operation is performed before the tip of the hammer HM returns to the position in the stationary state, at each point of the following, the integrator 57 that calculates the displacement of the hammer HM, and calculates the speed of the hammer HM. The integrator 56 is reset. In addition, at each of the above points a to c, the integrator 13 in FIG. 1 is also reset, so that the hammer speed signal HV is reset, and the apparatus enters a state of waiting for the next key pressing operation. The integrator 54 for calculating the displacement of the string STR may be reset in each of the cases a to c. However, by not intentionally performing this reset, it is possible to simulate a case where the hammer HM comes into contact by the next key depression operation before the vibration of the string STR is sufficiently attenuated. In this case, an error of the input signal or DC
Components are accumulated and output. Therefore, an integrator with a gain shown in FIG. 29 is used as the integrator 54. FIG.
0 illustrates the frequency characteristic S1 of the integrator with gain and the frequency characteristic S2 of the normal integrator. As shown in FIG. 30, the gain-added integrator has a lower gain in a low frequency region than a normal integrator. Therefore, by using an integrator with a gain as the integrator 54, it is possible to reduce the error or the accumulation of the DC component. In this case, the gain g is appropriately selected in consideration of the lowest note of the piano, a period at which a person can continuously press a key, and the like. It is also possible to initialize each integrator by determining the output level of the acceleration pickup 24 (FIG. 2), not in each of the cases a to c. For example, FIG. 31 shows the output signal waveform of the acceleration pickup 24 when the keyboard is continuously struck,
The reset operation may be performed at the time point when the zero cross point P0 (timing of transition from negative to positive) in this output signal waveform is detected.

Model <7>: Musical tone synthesizer considering damper In an actual piano, striking by a hammer and muting by a damper do not occur at the same time. Therefore, FIG.
In the configuration of the above, by performing simple operations such as switching of each parameter, that is, switching of the multiplication coefficient of each multiplier -1 / M, S, R, etc., the simulation of the hammering of the hammer and the damper using the same configuration are performed. Can simulate mute. The first specific example is shown in FIG. According to the configuration shown in FIG. 32, each parameter corresponding to the hammer and the damper is temporarily stored in the coefficient registers REG1 and REG2 when the power is turned on or when the parameters are operated and registered. Then, a key pressing operation is performed and the key switch signal KON / KO is input.
When the FF rises, the hammer parameters are read out from the coefficient register REG1, set in the respective sections of the excitation circuit 50, and a simulation of string striking is performed. Next, the pressed key is released and the key switch signal KON / KOFF
Falls, the parameter for the damper is read out from the coefficient register REG2 and set in each section of the excitation circuit 50, and the mute simulation by the damper is performed.

FIG. 33 shows a second specific example in which a damper is considered. This second example is in addition to the first example,
A low-pass filter 33D capable of controlling a pass characteristic is inserted in the loop circuit 30. When performing a simulation without muting, the low-pass filter 33D is invalidated, and the output of the adder 32 is supplied to the filter 33 as it is. On the other hand, when performing a simulation of playing the piano while muting, a predetermined filter coefficient is given to the low-pass filter 33D, and the acoustic loss given to the string STR by the damper is simulated. It is noted that the key release operation is performed as described above by the key switch signal KON / KOF.
In addition to the determination by F, the determination can be made by detecting that a negative pulse is generated from the acceleration pickup 24. In this case, after the setting of the parameters for the damper is completed, the negative pulse from the acceleration pickup 24 is directly used as a signal corresponding to the acceleration of the damper.
It may be input to the integrator 56 in FIG. Further, an integrated negative pulse may be input instead of the hammer speed signal HV. By doing so, mute is controlled according to the release touch, and various release feelings are realized.

[Configuration Related to Generation of Mechanical Noise] When a piano is played, a sound caused by vibration of a string and a sound generated by the vibration of the string propagated to a resonance system such as a soundboard or a frame are heard. In addition, so-called mechanical noise (mechanical noise) is heard. This mechanical noise is generated by the key hitting the pedestal or rubbing the action, which more or less characterizes the piano sound. Therefore, by simulating the mechanical noise, the reality (particularly, the attack portion) can be improved. This mechanical noise can be heard by our ears following a rather complicated path, such as the so-called direct sound, the sound that is filtered through the resonance system, and the sound that is injected from the piece into the string and emitted.

In order to simulate the occurrence of the mechanical noise, the output of the acceleration pickup 24 attached to the key is used. The signal MK corresponding to the acceleration applied to the pedestal when the key hits the pedestal is obtained by subtracting the offset OFFSET from the portion corresponding to the negative region in the output waveform of the acceleration pickup 24 shown in FIG. It is obtained by extraction using the configuration shown in FIG. The signal MK obtained in this way is A / D-converted, passed through a filter 4 (FIG. 1) that approximates the characteristics of the frame and soundboard, and added to the output from the tone synthesizer 15. In this case, as shown in FIG. 35, the signal M corresponding to each key
A / D conversion after adding K, MK,... Is economically realistic. However, when it is necessary to generate mechanical noise closer to that of an actual piano, the configuration shown in FIG. 36 is used. In this configuration, signals MK, MK,... Are added and output in octave units. Then, the addition results of the signals MK, MK,... For each octave are respectively converted into A / D converters 3-1 to 3-m (m is the number of octaves).
A / D-converted by the filter, passes through filters 4-1 to 4-m having different characteristics, and is added to a tone signal from a tone synthesizer.

In addition to the above mechanical noise, there is a string generated when a string is applied to the piano body. This is a piece,
Alternatively, the vibration of the frame is injected into the strings from the bridge, which is generated. This string also outputs the output of the filter 4 for generating mechanical noise to a position corresponding to a piece or a bridge in the musical tone synthesizer 15 (for example, FIG. 6).
The input can be reproduced by injecting it into the input terminal of the phase inversion circuit 34). In addition to inputting the signal obtained from the acceleration pickup 24 to the filter 4 as described above, a waveform corresponding to the impact of a hammer may be stored in a memory, and this waveform may be read and applied to the filter 4. Good. In addition, the output may be used by attaching a pressure sensor to the pedestal without using the acceleration pickup 24, or storing the waveform of the mechanical noise itself in the memory and adding it to the tone signal at the output stage. Good.

[Resonance system] The piano sound generally heard is
Not only those that are based on pure string vibrations, but also include synthetic sounds resulting from string vibrations being convolved with sound boards, frames, and the like. Therefore, it is conceivable that the tone signal obtained by the tone synthesizer 15 is input to the resonance system. The resonance system can be realized by a waveguide or a configuration in which a comb filter and an all-pass filter are combined. By the way, the resonance characteristics of a piano have a considerable difference depending on the position of a string to be sounded. Therefore, it is ideal to provide a resonance system for each tone synthesis unit corresponding to each string, but this is not realistic in view of the amount of calculation. Therefore, it is realistic to add a musical tone signal for 88 keys and use it as an input to the resonance system, and to prepare a resonance system at a rate of about one per octave. In this case, the characteristics of each resonance system are used according to the sound range. Further, the output of the resonance system may be fed back to the musical sound synthesizer 15 in order to simulate the connection between the string and the soundboard.

[Application to non-whole sound keyboard electronic musical instrument]
The above-described keyboard electronic musical instrument is basically a model of whole sound generation (for example, 88 keys, 88 sound sources). However, the present invention can also be applied to keyboard electronic musical instruments having different numbers of keys and sound sources, such as 16 sound sources with 88 keys and 32 sound sources with 88 keys. However, if the model is a whole-tone pronunciation, CP
It is not necessary for U1 to recognize the key code, the number of pronunciations, assign sound sources, etc.
U1 needs to make a key assignment. If the model is not a full-tone model, musical tones with different pitches are generated by the same sound source. In this case, it is necessary to switch the coefficient of the high-order FIR filter according to the pitch.
To make this possible, the result of calculating the filter coefficient corresponding to each key code is stored in the coefficient register in advance, and when a key-on event occurs, the CPU 1 stores the filter coefficient corresponding to the key code from the coefficient register. It is configured to supply to a sound source that performs readout sound generation. In this case, the order of the filter is higher for the lower strings, and there is a difference of several tens of times from the higher strings. Therefore, by changing the capacity of the coefficient register for each key code, the memory capacity can be saved.

[0047]

As described above, according to the present invention,
First and second at least delay means connected in a closed loop
In the invention according to claim 1, the excitation signal is supplied to the first closed loop means, the excitation signal is raised to a power, and the raised excitation signal is converted to the second excitation signal. And the supply means for supplying the excitation signal to the first and second closed loop means and the signal circulating through the first closed loop means. Second supply means for supplying power to the second closed loop means;
Since the output means for synthesizing the signal circulating in the first closed loop means and the signal circulating in the second closed loop means and outputting it as a musical tone signal is provided, it is possible to use a simple configuration to produce not only horizontal vibration but also vertical vibration. The effect of being able to simulate and synthesizing a musical tone that is closer to an actual piano sound, including a string sound, is obtained.

[Brief description of the drawings]

FIG. 1 is a block diagram illustrating a basic configuration of a keyboard electronic musical instrument according to an embodiment of the present invention.

FIG. 2 shows a key KEYj in the keyboard electronic musical instrument shown in FIG.
FIG. 3 is a diagram illustrating the configuration of FIG.

FIG. 3 is a block diagram showing a configuration of a portion related to key press detection in the keyboard electronic musical instrument shown in FIG.

FIG. 4 is a waveform diagram showing signal waveforms at various parts in FIG.

5 is a waveform chart showing signal waveforms when keys are pressed by various touches in the keyboard electronic musical instrument shown in FIG. 1.

6 is a block diagram showing a configuration example of a musical sound synthesizer 15 in the keyboard electronic musical instrument shown in FIG.

FIG. 7 is a diagram illustrating a non-linear conversion performed in the musical sound synthesizer 15 shown in FIG. 1;

FIG. 8 is a diagram illustrating a nonlinear conversion performed in the musical sound synthesizing unit 15;

9 is a block diagram showing another configuration example of a portion related to key press detection in the keyboard electronic musical instrument shown in FIG. 1.

FIG. 10 is a waveform diagram showing a vibration waveform appearing on a string of a piano.

FIG. 11 is a block diagram showing a configuration of a model <1>, which is another configuration example of the tone synthesis unit 15;

FIG. 12 is a diagram showing a string STR and a hammer HM of a piano simulated by the musical sound synthesizer 15;

FIG. 13 is a block diagram illustrating a configuration example of a first-order all-pass filter.

FIG. 14 is a block diagram illustrating a configuration example of a first-order all-pass filter.

FIG. 15 is a diagram illustrating a phase characteristic of a first-order all-pass filter.

FIG. 16 is a diagram illustrating phase characteristics of a piano string.

FIG. 17 is a block diagram illustrating a configuration example of a two-all-pass filter.

FIG. 18 is a block diagram illustrating a configuration example of a multi-order all-pass filter.

FIG. 19 is a block diagram illustrating a configuration example of a lattice type all-pass filter.

20 is a block diagram showing the configuration of the n-th stage in FIG.

FIG. 21 shows a high order F in a loop circuit 30 of the musical sound synthesizer 15.
FIG. 7 is a diagram illustrating an amplitude response when an IR filter is inserted.

FIG. 22 shows a low order F in a loop circuit 30 of the musical sound synthesizer 15.
FIG. 7 is a diagram illustrating an amplitude response when an IR filter is inserted.

FIG. 23 is a block diagram showing a modification of the musical sound synthesizer 15.

FIG. 24 is a block diagram showing a modification of the musical sound synthesizer 15.

FIG. 25 is a block diagram showing a modified example of the musical sound synthesizer 15.

FIG. 26 is a block diagram showing a configuration of a model <3>, which is another configuration example of the tone synthesis unit 15;

FIG. 27 is a block diagram showing a configuration of a model <4> which is a configuration example of the musical sound synthesis unit 15 according to the present invention.

FIG. 28 is a block diagram illustrating a configuration of a model <5>, which is another configuration example of the tone synthesis unit 15;

FIG. 29 is a block diagram showing a configuration of a gain-added integrator used in a model <6>, which is another configuration example of the musical sound synthesizer 15.

30 is a diagram illustrating a frequency characteristic of the integrator with gain shown in FIG. 29;

FIG. 31 is a waveform chart showing the operation of model <6>.

FIG. 32 is a block diagram illustrating a configuration of a model <7>, which is another configuration example of the musical sound synthesis unit 15.

FIG. 33 is a block diagram illustrating another configuration example of a model <7>.

34 is a waveform diagram illustrating an operation of generating mechanical noise in the keyboard electronic musical instrument shown in FIG.

35 is a block diagram showing a configuration of a portion related to generation of mechanical noise in the keyboard electronic musical instrument shown in FIG. 1.

36 is a block diagram showing another configuration of a portion related to generation of mechanical noise in the keyboard electronic musical instrument shown in FIG. 1.

FIG. 37 is a block diagram showing a configuration of a conventional keyboard electronic musical instrument.

[Explanation of symbols]

KEYj: Key, 24: Acceleration pickup, 13:
... Integrator, 15 ... Sound synthesizer, 30, 30H ... Loop circuit, 31, 35, 31H, 35H ... Delay circuit, 32,
36, 32H, 36H, 39H, 85 ... adder, 50 ...
Excitation circuit, 81, 83 ... multiplier.

Claims (2)

[Claims] 1. First and second closed loop means (30, 30H) in which at least delay means are connected in a closed loop; an excitation signal generating means (50) for generating an excitation signal; Supply means (32, 3) for supplying the excitation signal to the second closed loop means while supplying the excitation signal to the second closed loop means.
6, 32H, 36H, 81), and an output means (85) for combining a signal circulating through the first closed loop means and a signal circulating through the second closed loop means and outputting as a tone signal. An electronic musical instrument characterized by that: 2. First and second closed loop means (30, 30H) in which at least delay means are connected in a closed loop, an excitation signal generating means (50) for generating an excitation signal, and said first and second closed loop means First supply means (32, 36, 32H, 36H) for supplying to the second closed-loop means; and second supply for supplying a signal circulating through the first closed-loop means to the second closed-loop means. Means (39H, 8
3) and output means (85) for combining a signal circulating in the first closed-loop means and a signal circulating in the second closed-loop means and outputting as a tone signal. Musical instruments.