JPH0455893A - Electronic musical instrument - Google Patents

Abstract

PURPOSE:To additionally widen the dynamic range of timbres by changing the characteristics of a filter means in accordance with at least one of musical tone parameters to be inputted from an operating element. CONSTITUTION:The characteristics of the filter means 4 connected to the output circuit of a physical model sound source 2 are controlled in accordance with at least one of the musical tone parameters for driving the physical model sound source. Namely, the filter characteristics of the output circuit in addition to the musical tone parameters used heretofore in the conventional physical model sound source 2, such as bow pressure or bow speed are used also as the musical tone parameters. Then, the wider dynamic range of the timbres is obtd. and since the parameters for controlling the filter characteristics are formed in accordance with the musical tone parameters used heretofore with the physical model sound source, there is no need for preparing the fresh operating element. The wider changes of the timbres than heretofore are obtd. in this way by the operation meeting human sensation.

Description

[Detailed Description of the Invention] [Industrial Field of Application] This invention relates to an electronic musical instrument having a so-called physical model sound source that physically simulates the vibration system of a natural musical instrument using an electric circuit, and particularly relates to an electronic musical instrument having a so-called physical model sound source that physically simulates the vibration system of a natural musical instrument using an electric circuit. Regarding electronic musical instruments with a wider range.

[Prior Art] As an electronic instrument that generates the performance sound of a bowed string instrument such as a violin, a so-called physical model sound source, in which the mechanical vibration system of a bowed string instrument is physically simulated using an electric circuit, is used as a musical waveform signal forming means. It is known what it has. In such electronic musical instruments, bow pressure and bow speed are given as real-time parameters of a bowed string model (a physical model of a bowed string instrument) from an operator, and pitch information is given from a keyboard to control the sound source.

However, with conventional electronic musical instruments that only change the bow pressure and bow speed parameters of the sound source, it has not been possible to obtain a sufficient dynamic range of tone.

The present invention has been made in view of the drawbacks of the prior art, and it is an object of the present invention to provide an electronic musical instrument that uses a physical model sound source and has a wider dynamic range of timbre.

[Means for Solving the Problems] In order to achieve the above object, in the present invention, in an electronic musical instrument using a physical model sound source, the characteristics of the filter means connected to the output circuit of the physical model sound source are changed from those of the physical model sound source. Control is performed based on at least one of the musical tone parameters for driving.

[Functions and Effects] According to the above configuration, in addition to the musical tone parameters used in conventional physical model sound sources, such as bow pressure and bow speed, the filter characteristics of the output circuit are also used as musical tone parameters, thereby creating a wider range of tones. We were able to obtain a dynamic range of Furthermore, since the parameters for controlling the filter characteristics are created based on the musical tone parameters used in the conventional physical model sound source, there is no need to prepare new operators. As a result, it was possible to obtain a wider range of tonal changes than before by using operations that matched the human senses without changing conventional operations.

[Examples] Hereinafter, the present invention will be explained in more detail based on Examples.

FIG. 1 shows a block configuration of an electron source stringed instrument according to an embodiment of the present invention.

In the figure, the real-time operator 1 includes a pressure-sensitive slide type performance operator 11 consisting of a slide volume equipped with a pressure-sensitive means. Note that the performance operator may be a joystick mechanism or a mouse mechanism equipped with a pressure sensitive means. Position information obtained by operating the performance controller 11 is provided by A/D.
The data is converted into velocity data via the converter 12 and the position/velocity conversion arithmetic circuit 13. This speed data is input to the bowed string model sound source 2 as 5th speed data. Further, pressure data from the pressure sensing means of the performance operator 11 is inputted to the sound source 2 as bow pressure data.

FIG. 2 shows an example of the circuit of the bowed string model sound source 2.

In the figure, numerals 202 and 203 indicate adders and correspond to the string points. 204 and 205 indicate multipliers, which correspond to the string ends on both sides of the string point. Adder 202, delay circuit 206, low pass filter 207, attenuator 208 and multiplier 204
The closed loop consisting of corresponds to the string on one side of the chord point. Similarly, adder 203, i! A closed loop consisting of extension circuit 209, low-pass filter 210, attenuator 211, and multiplier 205 corresponds to the string on the other side of the chord point. The delay time of each closed loop corresponds to the resonant frequency of the string on the corresponding side. 212 indicates a nonlinear function generator.

This nonlinear function generator 212 adds a signal corresponding to the bow speed to a signal obtained by combining the outputs of closed loops on both sides of the chord point in an adder 213, and further adds a signal corresponding to the bow speed to a signal from a fixed hysteresis low-pass filter 214. A signal obtained by adding a signal obtained by multiplying Kane G by a multiplier 215 is input. Further, hysteresis control of the nonlinear function generator 212 is performed using a signal corresponding to the bow pressure.

The musical waveform signal output from the sound source 2 is taken out from the output end of the delay circuit 206 via the filter circuit 4 which is made up of a cascade circuit of a low pass filter (LPF) 41 and a bypass filter (HPF) 42.

Referring to FIG. 1, pitch data corresponding to a musical scale is further input to the sound source 2 by operating a pitch specifying means 3 consisting of a voltament bar, a keyboard, or the like. The sound source 2 forms a musical tone waveform No. 48 based on the 5th speed data and bow pressure data output from the operator 1 and the pitch data output from the pitch specifying means 3.

This musical waveform signal is input to the sound system 5 via the filter circuit 4, and is emitted as an electronic sound.

The filter characteristic control circuit 6 controls the speed/speed as shown in FIG.
Equipped with an LPF coefficient conversion table, filter characteristic control data (L
PF coefficient) is created and this control data is sent to the filter circuit 4.
supply to. The filter circuit 4 is controlled by its transfer characteristic or by the control data. In the table of FIG. 3, as the operating speed of the operator increases, the LPF coefficient also increases, and the tone becomes brighter.

FIG. 4 shows a circuit example of the LPF 41. LP like this
The F circuit is described in Japanese Patent Application Laid-open No. 81-18212 and Japanese Patent Application No. 1-262265, which is an earlier application filed by the present applicant.

The transfer function H(z) of this filter is expressed as follows: α is the multiplication coefficient of multiplier M, and z-1 is a symbol representing data one sampling frequency before. Also, the cutoff frequency f. is α is 1
If it is sufficiently smaller than (where fS is the sampling frequency), it is expressed as follows. That is, this digital order low-pass filter has almost the same frequency characteristics as an analog filter expressed by the transfer function H(s) or H(S)-s+a (where a=αfs).

Also, by controlling the multiplication coefficient α of the multiplier M,
The transfer function H(z) can be controlled. Therefore, when using a digital LPF as shown in FIG. 4 as the filter circuit 4 in FIG.
may be configured to create data indicating the multiplication coefficient α as the control data.

FIG. 5 is a block diagram of a control mechanism of an electronic musical instrument including the filter characteristic control circuit.

The key switch circuit 51 detects a pressed key on the keyboard 52, and a key code (K) representing the key.
CD). Key coat (KCD) output from the key switch circuit 51, pressure-sensitive slide type performance controller 5
Pressure data and position data output from the pressure detection circuit 54 and the position detection circuit 55 in response to the operation No. 3 are input to the CPU 57 via the lotus line 56. CPU
57 reads out necessary data from a ROM 58 that stores data necessary for arithmetic processing such as each routine program and the table shown in FIG. Parameters (5 pressure data, 5 speed data, 'pitch data) and parameters for filter characteristic control (LPF coefficient) are calculated and sent to the physical model sound source 2 and filter circuit 4.

The timer 60 interrupts the main routine of the program by the CPU 57 at a fixed cycle of about several milliseconds.

The following registers are set in the RAM 59. In the following, it is assumed that the register salt and the data stored in the register are represented at the same level.

KCD: Key code register PH1: Pressure data register VEL: Velocity data register pos: Position data register X: Previous position data register DIF: Position difference data register LPF: LPF coefficient data register
The main routine processing executed by U57 (FIG. 5) is shown. The CPU 57 starts operating when the power is turned on.

First, in step 61, each register is initialized. Also,
Set each parameter to a predetermined initial value. Next, cycle 3] processing consisting of key-on event detection processing (step 62), key-off event detection processing (step 64), and other processing (step 66) is repeated.

During this cycle process, when a key-on or key-off event is detected, key-on or key-off event processing, which will be described later, is executed. Further, for such a main routine, an interrupt routine (described later) is executed at a constant cycle by the timer 60.

When a key is pressed on the keyboard 52 of FIG. 5, the CPU 57 detects this in step 62 during the circulation process and executes key-off event processing in step 63.

FIG. 7 shows details of this key-on event processing. First, the key code of the key pressed on the keyboard 52 is imported from the key switch circuit 51 (Fig. 5) and stored in the RAM.
59 (step 71). Next, this key code (KC
D) After assigning the sound to the sound generation channel CH of the physical model sound source 2 (FIG. 5) (step 72) and transmitting the key code (KCD) and key-on signal to the corresponding channel CM of the sound source 2 (step 73), the 6th Return to step 64 in the figure.

CPtJ57 has 1mi&! key on the keyboard 52 in FIG. If so, step 64 during the cyclic process
The stiffness is detected at step 65, and key-off event processing is executed at step 65.

FIG. 8 shows details of this key-off event processing. First, the key code of the key released by the key photo 52 is fetched from the key switch circuit 51 and stored in the key code register (KCD) set in the RAM 59 (step 81). Next, among the sound generation channels of the physical model sound source 2, the channel to which this key code (KCD) sound is assigned is detected (step 82). Step 83
Then, it is determined whether or not the corresponding channel CH exists. If there is a corresponding channel CH, the corresponding channel CH of sound source 2
After transmitting the key-off signal to (step 84), if there is no corresponding channel, nothing is done and the process returns to step 66 in FIG.

FIG. 9 shows an interrupt routine that interrupts the main routine at regular intervals using a fixed clock.

First, the pressure value obtained by the pressure detection circuit 54 in FIG. 5 is sent to the pressure data register PR5, and the position data obtained by the position detection circuit 55 is sent to the position data register PO5.
(step 91). Next, it is determined whether there is a key-on signal (step 92), and if there is a key-on signal,
Furthermore, it is determined whether the pressure data PR5 is 0 or not (
Step 93). If there is no key-on signal in the judgment of step 92, and if there is no key-on signal in the judgment of step 93, the pressure data PR
, S is 0, the interrupt is canceled and the process returns to the original routine.

On the other hand, if it is determined in step 92 that there is a key-on signal, and the pressure data PR3 is not O as determined in step 93, the difference between position PO5 and the previous position X, which is the position at the time of the previous interrupt processing, is calculated; This is stored in the position difference data register DTF (step 94). Next, the speed table and LPF table are accessed using this position difference data DIF as a parameter, and the obtained values are used as the speed VEL.
and stored in the LPF coefficient data register LPF (step 95). Furthermore, the previous position data
After updating the oS (Step 96), storing the velocity VEL in the velocity data register VEL (Step 97), transferring the pressure data PR3 and velocity data VEL to the sound source 2, and transferring the LPF coefficient data LPF to the filter circuit 4 (Step 98). , cancel the interrupt and return to the original routine.

As described above, by changing the characteristics of the filter circuit 4 according to an external signal and changing the characteristics of the filter circuit 4 according to the speed data output from the controller 1, sound is emitted through the sound system 5. It was possible to make the range of change in the timbre of musical tones (dynamic range) wider than when the characteristics of the filter circuit 4 were fixed. Furthermore, since the characteristics of the filter circuit 4 are controlled by the output (speed data) from the operator 1, there is no need to provide a new operator. Therefore, it was possible to obtain timbre changes with a larger dynamic range using the same operations as conventional bowed string electronic instruments.

[Modifications of Embodiments] The present invention is not limited to the above-described embodiments, and can be implemented with appropriate modifications.

For example, in the above embodiment, a slide-type performance controller equipped with a slide polyolium was used as the performance controller, but any controller that outputs pressure, position, or velocity data corresponding to real-time operation may be used. may be other types of performance controls.

Further, the input source of pressure data and position or velocity data is not limited to the operator. The input may be made from a computer system or the like other than the operator.

In the above-described embodiment, an example is shown in which a bowed string model is used as a sound source, but it is also possible to use a pipe model, or even an FM sound source other than a physical model sound source.

The data conversion table can be selected as appropriate. For example, the speed/LPF coefficient table is shown in Figure 1 or Figure 11.
It may also represent a function as shown in the figure.

Furthermore, the data conversion table does not need to include all the table data, but may include a portion of the table data, with interpolation performed between them.

Data conversion may be obtained by calculation or the like instead of using a table.

[Brief explanation of the drawing]

1 is a block configuration diagram of an electronic musical instrument according to the present invention; FIG. 2 is a circuit diagram of the bowed string model sound source and filter circuit portion in FIG. 1; FIG. 3 is a filter characteristic control circuit in FIG. 1. A graph showing a speed/LPF coefficient data conversion function which is a human output characteristic, FIG. 4 is a circuit diagram showing an example of the filter circuit in FIG. 2, and FIG. 5 is a graph showing an example of the filter circuit of the present invention. A configuration diagram of an electronic musical instrument using a CPU. Figure 6 is a flowchart of the main routine of the program executed by the CPU in Figure 5. Figure 7 is a flowchart at key-on. Figure 8 is a flowchart at key-off. Flow Diagram FIG. 9 is a flow diagram of a timer interrupt routine, and FIGS. 10 and 11 are graphs similar to FIG. 3 showing other examples of data conversion functions, respectively. 1: Operator 2: Physical model sound source 4: Filter circuit 6: Filter characteristic control circuit

Claims (1)

[Claims] (1) A performance operator that generates one or more musical tone parameters, a closed loop means including a delay means, and a waveform signal that receives the musical tone parameters input from the operator and the waveform signal taken out from the closed loop means. a musical waveform signal forming means comprising a drive signal generating means that changes the signal according to the musical tone parameters and supplies the changed signal to the closed loop means; and a filter means whose characteristics can be changed, which is connected to an output circuit of the musical waveform signal forming means. An electronic musical instrument comprising: filter characteristic control means for changing characteristics of the filter means based on at least one musical tone parameter input from the operator.