JP3223889B2 - Music sound synthesizer, music sound synthesis method, and storage medium - Google Patents

Description

DETAILED DESCRIPTION OF THE INVENTION

[0001]

BACKGROUND OF THE INVENTION 1. Field of the Invention The present invention relates to a musical tone synthesizing apparatus, a musical tone synthesizing method, and a storage medium for synthesizing a musical tone waveform having an effect by executing an effect algorithm program.

[0002]

2. Description of the Related Art Conventionally, when an effect is applied to a polyphonic tone by a tone synthesizer,
First, a musical tone waveform corresponding to each note constituting the target polyphonic musical tone is generated, and then, the generated musical tone waveforms are mixed to generate a polyphonic musical tone waveform. For example, DSP (Digita
l Signal Processor), and the effect is applied to the polyphonic musical tone by executing a predetermined effect algorithm program in the effect circuit.

[0003]

However, in the above-described conventional tone synthesizing apparatus, various effects can be obtained for each key-depressed tone, that is, for each note constituting a polyphonic tone, and individually for each note (for example, unique information). It is difficult to give an effect based on the pitch and timbre, and to perform a precise timbre or timbre control including the sound effect.

The present invention has been made in view of this point, and provides an effect based on information unique to each note constituting a polyphonic musical tone, thereby expanding the range of the effect and improving the effect. It is an object of the present invention to provide a musical tone synthesizing apparatus, a musical tone synthesizing method, and a storage medium capable of performing minute tone control individually for each note including the musical note.

[0005]

In order to achieve the above object, a musical tone synthesizing apparatus according to claim 1 includes a musical tone specifying means for specifying a tone color and a pitch of a musical tone to be generated. By executing a basic waveform generating means for generating a basic tone waveform corresponding to a tone color and a pitch, and executing an effect algorithm program, the generated basic tone waveform is generated.
On the other hand, there is provided an effect imparting means for imparting an effect including a process of delaying the basic musical tone waveform, and a control means for controlling the effect imparting means to execute an effect algorithm program corresponding to the designated timbre. and, the effect algorithm, in order to impart a electromagnetic pickup-specific effect of the electric guitar, the basic music that has been input
An algorithm for changing a delay amount of delaying a sound waveform according to an assumed picking position of the electric guitar and an assumed position of the electromagnetic pickup, wherein the picking position and the pickup position are the designated positions. It is characterized in that control is performed in accordance with the pitch of a musical tone.

[0006] A third aspect of the present invention provides a musical tone synthesizing method.
A tone designation step of designating a tone and a pitch of a tone to be generated by the tone designation means; a basic waveform generating step of generating a basic tone waveform corresponding to the designated tone and pitch;
By executing the effect algorithm program, for the generation elementary tone waveform, the fundamental tone waveform
An effect applying step of applying an effect including a process of delaying, and a control step of controlling the effect applying step to execute an effect algorithm program corresponding to the specified tone color, wherein the effect algorithm includes: In order to provide the effect unique to the electromagnetic pickup of the electric guitar, the delay amount for delaying the input basic musical sound waveform is changed according to the assumed picking position of the electric guitar and the assumed position of the electromagnetic pickup. Wherein the picking position and the pickup position are controlled in accordance with the pitch of the specified musical tone.

Further, a storage medium according to a fifth aspect of the present invention provides a tone specifying module for specifying a tone and a pitch to be generated by a tone specifying means, and a basic tone waveform corresponding to the specified tone and pitch. a basic waveform generating module, by executing the effect algorithm program, for the generation elementary tone waveform, the basic music
An effect imparting module that imparts an effect including a process of delaying a sound waveform, and a control module that controls the effect imparting module to execute an effect algorithm program corresponding to the specified timbre, wherein the effect algorithm is Input the basic music to provide an effect peculiar to the electromagnetic pickup of an electric guitar.
An algorithm for changing a delay amount of delaying a sound waveform according to an assumed picking position of the electric guitar and an assumed position of the electromagnetic pickup, wherein the picking position and the pickup position are the designated positions. It is characterized in that control is performed in accordance with the pitch of a musical tone.

[0008]

[0009]

[0010]

Further, according to a second aspect of the present invention, there is provided a musical tone synthesizing apparatus for specifying a tone color and a pitch of a tone to be generated, and a basic tone waveform corresponding to the specified tone color and pitch. By executing the generated basic waveform generating means and the effect algorithm program, the effect applying means for applying the effect to the generated basic musical sound waveform, and executing the effect algorithm program corresponding to the specified tone color. and control means for controlling said effect imparting means, the effect algorithm generates a random signal, by controlling the cut-off frequency based on the random signal, 該Ka cutoff frequency is randomly controlled
Band limiting the generated basic musical sound waveform by a band-pass filter, and injecting the band-limited musical sound waveform into a loop in which at least the amount of delay is controlled according to the pitch. .

[0012]

Further, in the musical tone synthesizing method according to the present invention, a musical tone specifying step of designating a timbre and a pitch of a musical tone to be generated by the musical tone specifying means, and a basic musical tone corresponding to the designated timbre and pitch. Executing a basic waveform generating step of generating a waveform and an effect algorithm program corresponding to the designated tone color by executing the effect algorithm program, thereby executing an effect applying step of applying the effect to the generated basic tone waveform. and a controlling process of controlling the effect imparting process to the effect algorithm generates a random signal, by controlling the cut-off frequency based on the random signal, orchids 該Ka cutoff frequency
The band is limited to the generated basic musical sound waveform by a dam-controlled bandpass filter, and the band-limited musical sound waveform is injected into a loop in which at least a delay amount is controlled according to the pitch. Features.

[0014]

Further, a storage medium according to a sixth aspect of the present invention provides a tone specifying module for specifying a tone and a pitch to be generated by a tone specifying means, and a basic tone waveform corresponding to the specified tone and pitch. A basic waveform generating module that executes the effect algorithm program, thereby executing an effect applying module that applies the effect to the generated basic musical tone waveform, and an effect algorithm program that corresponds to the specified timbre. and a control module for controlling the effect imparting module, the effect algorithm generates a random signal, by controlling the cut-off frequency based on the random signal, 該Ka Tsu <br/> off frequency is randomly controlled Bandpass filter
The band of the generated basic musical tone waveform is limited, and the band-limited musical tone waveform is injected into a loop in which at least the delay amount is controlled according to the pitch.

[0016]

Embodiments of the present invention will be described below in detail with reference to the drawings.

FIG. 1 is a block diagram showing a schematic configuration of a musical sound synthesizer according to an embodiment of the present invention.

As shown in FIG. 1, the musical tone synthesizing apparatus according to the present embodiment has a keyboard 1 for inputting pitch information and for instructing the sound generation, and a keyboard 1 for inputting various information including designation of a tone color. Panel switch 2 with multiple switches
And a key press detection circuit 3 for detecting a key press state of each key of the keyboard 1.
A switch detection circuit 4 for detecting the pressed state of each switch of the panel switch 2, and a CPU for controlling the entire apparatus
5, a ROM 6 for storing a control program and table data executed by the CPU 5, a RAM 7 for temporarily storing performance data, various input information, calculation results, and the like.
The timer 8 measures the interrupt time and various times in the timer interrupt processing, and displays various information such as a large liquid crystal display (LCD) or a CRT (Cathode).
Ray Tube) display and light emitting diode (LE)
D) or the like, a floppy disk drive (FDD) 10 for driving a floppy disk (FD) 20 as a storage medium, and a hard disk (FD) for storing various application programs and various data including the control program. A hard disk drive (HDD) 11 for driving various types of application programs and various data including the control program, and a CD-ROM for driving a compact disk-read only memory (CD-ROM) 21.
Drive (CD-ROMD) 12 and external MID
M for inputting an I (Musical Instrument Digital Interface) signal or outputting it as a MIDI signal to the outside
Via the IDI interface (I / F) 13 and the communication network 101, for example, the server computer 1
02, a communication interface (I
/ F) 14, a voice parameter memory 15 for storing voice parameters to be described later, and a tone generator 16 for converting performance data input from the keyboard 1 or preset performance data into musical tone signals and imparting various effects. And a DAC (Digi) called a so-called CODEC for converting a digital tone signal from the tone generator 16 into an analog tone signal.
(Tal-to-Analog Converter) 17 and a sound system 18 such as an amplifier or a speaker for converting a tone signal from the DAC 17 into sound.

The above components 3 to 17 are interconnected via a bus 19, a timer 8 is connected to the CPU 5,
Another MIDI device 100 is connected to the MID II / F 13, a communication network 101 is connected to the communication I / F 14, a DAC 17 is connected to the tone generator 16,
A sound system 18 is connected to 17.

As described above, the hard disk of the HDD 11 can also store the control program executed by the CPU 5, and when the control program is not stored in the ROM 6, the control program is stored in the hard disk. By reading into RAM7, ROM
6 can cause the CPU 5 to perform the same operation as when the control program is stored. This makes it easy to add a control program, upgrade a version, and the like.

CD-ROM of CD-ROM drive 12
The control program and various data read from 21 are
The data is stored on the hard disk in the HDD 11. This makes it possible to easily perform new installation, version upgrade, and the like of the control program. In addition to the CD-ROM drive 12, a device for utilizing various forms of media, such as a magneto-optical disk (MO) device, may be provided as an external storage device.

As described above, the communication I / F 14 is connected to a communication network 101 such as a LAN (Local Area Network), the Internet, or a telephone line.
Via the communication network 101, it is connected to another computer on the network 101, such as a server computer 102. When the above programs and various parameters are not stored in the hard disk in the HDD 11, the communication I / F 14 is used to download the programs and parameters from the server computer 102. The client computer (in this embodiment, the tone synthesizer) is connected to the server computer 102 via the communication I / F 14 and the communication network 101.
Send a command to request the download of programs and parameters. The server computer 102 receives the command, distributes the requested program or parameter to the computer via the communication network 101, and the computer receives the program or parameter via the communication I / F 14 and The download is completed by accumulating the data on the hard disk.

In addition, an interface for directly exchanging data with an external computer or the like may be provided.

The voice parameter memory 15 includes, for example, ROM, RAM, FDD, HDD, CD-ROMD, M
Any storage medium such as an O drive may be used. Tone data library stored in the voice parameter memory 15 (voice data and DSP
Program, etc.)
The voice parameter memory 15 varies depending on the storage medium that forms the voice parameter memory 15.
5 is composed of a RAM, usually a non-volatile storage medium (the ROM 6, FD 20, HD, CD-RO
M21, MO, etc.) and read out as appropriate. Also,
It may be placed on the communication network 101, that is, on the server computer 102 or another client terminal.

When a CPU having a high processing speed is used as the CPU 5, some or all of the functions of the sound source 16 may be transferred to the CPU 5 to reduce the load on the sound source 16. Good. In this case, DAC1
7 receives a musical tone waveform from the CPU 5 via the bus 19.

The apparatus to which the present invention is applied is not limited to a typical form of a musical sound synthesizer in an electronic musical instrument, but may be amusement equipment such as a game machine or a karaoke apparatus, various home appliances such as a TV apparatus, or a PC. Computer devices and systems to be used.

FIG. 2 shows a memory map of various memories, (a) shows a memory map of the ROM 6, (b) shows a memory map of the RAM 7, and (c) and (d) show memory maps of the voice parameter memory 15. 3 shows a memory map.

The ROM 6 has a CPU as shown in FIG.
The control program executed by the CPU 5 is stored in advance in a CPU program area, and the table data and various controls other than the control program are stored in the fixed data area. Data and the like are stored in advance.

The RAM 7 has a CPU as shown in FIG.
The CPU working area is composed of a working area (area) and a plurality of buffer areas. As described above, performance data, various input information, calculation results, and the like are temporarily stored in the CPU working area.

The buffer area includes a first wave parameter buffer (WAVEPARBUF1) for storing a first-system wave parameter (a parameter or a parameter group that determines the waveform of music to be pronounced), and a first-system effect (effect) parameter. , A second effect parameter buffer (WAVEPARBUF2) for storing second-system wave parameters, and a second effect parameter buffer (WAVEPARBUF2) for storing second-system effect parameters. EFCTPARBUF2) and changes the contents of the wave parameter buffer of the system corresponding to this key-on based on information such as the key code and touch (eg, velocity) corresponding to this key-on when the user issues a sounding instruction (key-on). Key-on to remember And Bubaffa (KEYONWAVEBUF), likewise key-on effect buffer contents of the effect parameter buffer system corresponding to the key-on, and stores the changed based on the key code and the touch information such as corresponding to the key-on (KEYO
NEFCTBUF). Here, each of the wave parameter buffer and the effect parameter buffer is configured to have two similar buffers. The musical tone synthesizing apparatus according to the present embodiment has two systems (the system corresponds to “tone color”, This is because the same timbre can be assigned to each system, so that the musical tone waveform of (not using the term “timbre”) can be synthesized.

The voice parameter memory 15 has a voice data area ((c)) for storing a plurality of voice data (VOICE1-i) and a D for storing a plurality of DSP programs.
SP program area ((d)). The DSP program includes a wave program (WAVEPRG1-j) for generating a basic tone waveform and an effect program (NO) for giving an effect to the basic tone waveform.
Therefore, the DSP program area includes a wave program area and an effect program area.

Since a plurality of (i) voice data stored in the voice data area have the same data format, the first voice data will be described below in detail. I do.

The voice data includes voice name data (VOICE NAME) indicating the name of the voice, as shown in FIG.
And wave parameter data (WAVEPAR) used to generate a musical sound waveform corresponding to the voice, effect parameter data (NOTEFCTPAR) used to add an effect corresponding to the voice, and, for example, Common parameter data (CO
MMONPAR). Note that the common parameter data may be incorporated in the wave parameter data or the effect parameter data, and the storage area may not be provided separately.

The wave parameter data includes wave algorithm data (WAVEALG) for designating a wave algorithm program for generating a basic tone waveform corresponding to the voice, and various wave parameters (WAVEPAG) used in the designated wave algorithm program.
R1-m). The effect parameter data includes an effect algorithm data (NOTE) that specifies an effect algorithm program for generating an effect to be applied to the basic tone waveform corresponding to the voice.
FCTALG) and various effect parameters (NOTEFC) used in this specified effect algorithm program.
TPAR1-n).

In the wave program area,
Various wave programs (WAVEPRG1-j) are stored, and each wave program is specified by the wave algorithm data. Various effect programs (NOTEFCTPRG) are stored in the effect program area.
1-k) is stored, and each effect program is specified by the effect algorithm data.

When the user specifies a timbre for each system, wave parameter data (WAVEPAR) corresponding to the specified timbre is selected, and a wave program specified by the wave algorithm data among the wave parameter data is selected. Are transferred to the program register of the system (system corresponding to the designated tone color) of the sound source unit 16, and various wave parameters (WAVEPAR1-m) of the wave parameter data correspond to the first or second program corresponding to the system. 2 is stored in the second wave parameter buffer. Similarly, the effect parameter data (NOTEFCTPAR) corresponding to the specified tone is selected, and the effect program specified by the effect algorithm data among the effect parameter data is stored in the program register of the sound source unit 16 in the relevant system. The effect parameters are transferred and various effect parameters (NOTEFCTPAR1-n) of the effect parameter data are stored in the first or second effect parameter buffer corresponding to the system.

When the user performs a key-on (pronunciation instruction), the contents of the wave parameter buffer of the system corresponding to the key-on are changed based on the data such as the key code and the touch of the key corresponding to the key-on. After being stored in the key-on wave buffer, it is transferred to a parameter register (register for storing wave parameters) of the sound source unit 16 of the relevant system. Similarly,
After the contents of the effect parameter buffer of the system corresponding to this key-on are changed based on the key code and touch data of the key corresponding to this key-on, and stored in the key-on effect buffer, the contents of the system of the sound source unit 16 are changed. It is transferred to a parameter register (a register for storing effect parameters).

FIG. 3 is a block diagram showing the structure of the sound source section 16. As shown in FIG.

As shown in the figure, the sound source section 16 is provided with a wave DSP (WAVEDS) for generating a basic tone waveform of the first system.
P1) 161 and an effect DSP (EFCTDSP1) 162 for giving an effect to the basic musical sound waveform, and a wave DSP (WAVEDSP2) 16 for generating a basic musical sound waveform of the second system.
3 and an effect DSP (EFCTDSP2) 164 for giving an effect to the basic musical tone waveform.

The wave DSP 161 is interconnected with the bus 19 and is connected to the effect DSP 162 for giving an effect to the generated basic musical sound waveform. Similarly, the wave DSP 163 is interconnected with the bus 19, and applies an effect to the generated basic musical sound waveform.
4 is connected.

The effect DSP 162 is connected to the bus 1
9 and to the DAC 17. The effect DSP 164 is connected to the bus 19 and to the effect DSP 162. The reason why the output of the effect DSP 164 is input to the effect DSP 162 is that the effect DSP 162
In addition to the function of imparting an effect to the basic tone signal generated by the above, an effect is imparted to the first series of tone signals to which this effect has been applied and the output signal from the effect DSP 164, ie, the basic tone signal generated by the wave DSP 163. The added second tone signal is combined (added) to the DAC 17
This is because it has the function of outputting to Of course, the first and second series of effect DSPs 162 and 164 are both DSPs having only a function of giving an effect to the basic tone signal, and an adder for adding the output signals from the effect DSPs 162 and 164 is separately provided. Is also good.

Thus, each of the DSPs 161 to 164 has:
Although the functions are different from each other, each of the DSPs 161 to 1
There is no difference in the hardware configuration between the 64. Therefore, hereinafter, the details of the hardware will be described on behalf of one DSP.

FIG. 4 shows the four DSPs 161 to 164.
3 is a block diagram showing details of hardware of any one DSP 16k (k indicates an integer value of any one of 1 to 4). FIG.

In the figure, the DSP 16k is connected to a bus 19
A data bus interface 16k1 for interfacing with a data bus, and the DSP 16
k), a parameter register (PARREG) 16k2 for storing parameters required for signal processing at k, and a control program (D) for describing the procedure of signal processing performed by the DSP 16k.
Program memory (PRGME) for storing SP programs
M) 16k3 and mainly the parameter register 16k2
With reference to the parameters stored in the
A signal processing operation unit 16k4 that performs various signal processing operations based on the control program stored in 6k3, and a case where a signal serving as the basis is required when the signal processing operation unit 16k4 performs a signal processing operation (for example, Effect DS
P162 uses the basic tone signal generated by the wave DSP 161 for the signal processing), a signal input unit 16k5 for inputting a signal from the outside and outputting it to the signal processing operation unit 16k4, It is mainly composed of a signal calculation RAM 16k6 for temporarily storing the calculation results and the like, and a signal output unit 16k7 for temporarily storing the signal processing results after the signal processing calculation and outputting them to the outside at a predetermined timing. .

The parameter register 16k2 and the program memory 16k3 are connected to the data bus interface 1
6k1 and the signal processing operation unit 1
6k4. The signal processing operation unit 16k4 further includes a signal input unit 16k5, a signal operation RAM 1
6k6 and the signal output unit 16k7 are connected.

When the DSP 16k is, for example, the effect DSP 164, the signal input section 1645 is connected to the data bus interface 1641 and the wave DSP.
163, and the signal output unit 137
647 is connected to the data bus interface 1641. The signal input unit 1645 and the signal output unit 16
The operation of 47 and the procedure of exchanging signals can be set as appropriate in accordance with the system, in accordance with instructions to software or a DSP or setting parameters.

FIG. 5 is a diagram for explaining the tone generation processing executed by the tone generator 16.

In the figure, for example, when the user selects a timbre (voice) by operating the panel switch 2, the voice select information is transmitted to the channel (C).
H) / Sent to the voice parameter designation unit 51. Similarly, when a note event occurs when the user presses the keyboard 1, the note event information is transmitted to the channel / voice parameter designation unit 51. Here, the channel / voice parameter designating section 51 is constituted by the CPU 5, for example. That is, the processing performed by the channel / voice parameter specifying unit 51 is performed by the CPU.
5 is a part of the processing performed.

Channel / voice parameter designating section 51
First specifies the system for which key-on / off was instructed,
Next, when the key-on is instructed, the sounding channel is designated, and when the key-off is instructed, the sounding channel is designated. Then, based on the input voice select information and note event information, key information (for example, key on / off, key code and touch) is generated, and the key-on wave buffer and key-on effect buffer stored as described above are stored. Is read out. Further, the key information and the read timbre parameters (ie, the contents of the key-on wave buffer) are stored in the designated channel of the designated wave DSP 161 or 163.
And the effect DS of the specified system
The key information and the read effect parameter (that is, the contents of the key-on effect buffer) are set to the designated channel of P162 or 164.

Accordingly, the wave D of the designated system
The SP 161 or 163 generates a basic tone waveform wavek (k is an integer from 1 to n) of the designated channel, and the effect DSP 162 or 1
64. The effect DSP 162 or 1 of the designated system is applied to this basic musical sound waveform wavek.
The effect is given by 64, and the musical sound waveform choutk to which the effect is given is output. Then, each musical sound waveform output chouk (k = 1,..., N) of each system is added by the effect DSP 162 as described above,
Output to AC17.

In this embodiment, eight sound waveforms (n = 8) of the same tone color and different pitches for one system can be synthesized with the sound waveforms of all two systems. In other words, when different timbres are assigned to the respective systems, up to eight tone waveforms of two timbres can be synthesized for each timbre.
It is configured so that a maximum of 16 tones of musical tone waveforms can be synthesized.

FIG. 6 is a graph showing the basic tone waveform wavek and tone waveform output c in one channel _k.
FIG. 11 is a diagram for describing a method of generating a hook in more detail.

In this embodiment, one sound is composed of a plurality of (for example, four) elements. of course,
Depending on the sound, there may be a case where all the elements are not used and some of the elements are used.

In the figure, the number of elements used, the waveform generation method for each element, the set values of filter coefficients and amplitudes, and the mixing coefficients are input as the timbre parameters. A basic musical tone waveform is generated for each element based on the tone color parameter, weighted by a mixing coefficient, and then mixed to generate a basic musical tone waveform wavek. Similarly, the effect DSP 162 or 16
4, an effect algorithm and an effect parameter corresponding to this algorithm are input as the effect parameters, and based on these effect parameters, an effect is given to the generated basic tone waveform wavek, and the tone waveform output is generated. A chouk is generated.

The control processing executed by the tone synthesizer constructed as described above will be described in detail below with reference to FIGS.

FIG. 7 is a flowchart showing the procedure of a main routine executed by the tone synthesizing apparatus of this embodiment, in particular, by the CPU 5.

In the figure, first, initialization such as clearing of the RAM 7 is performed (step S1).

Next, an operation performance event detection process for detecting whether or not the keyboard 1 and the panel switch 2 are operated by the user to generate the operation performance event is performed (step S2). It is determined whether an event has occurred (step S3).

When a tone color designation event has occurred in step S3, a tone color parameter setting process subroutine described later with reference to FIG. 8 is executed (step S4).
If the tone designation event does not occur, step S4
Skip to step S5.

In step S5, it is determined whether or not a key-on event has occurred by the detection processing in step S2. If a key-on event has occurred, a sounding instruction processing subroutine described later with reference to FIG. 9 is executed (step S6). On the other hand, when the key-on event does not occur, the process skips step S6 and proceeds to step S7.

In step S7, for example, after executing a mute process when a key-off event is detected, and other processes such as an event handling process when another event is detected and a tone data editing process, Step S
Returning to step 2, the above processing is repeated.

FIG. 8 is a flowchart showing the detailed procedure of the tone color parameter setting processing subroutine in step S4.

In the figure, first, a voice number and a series number are determined from a tone color designation event, and RA and RA are respectively determined.
It is stored in the predetermined areas VNO and MLTNO of M7 (step S11). Hereinafter, the content stored in area VNO is referred to as voice number VNO, and the content stored in area MLTNO is referred to as sequence number MLTNO.

Next, in the voice data VOICE (VNO) specified by the voice number VNO, that is, in one of the voice data VOICE1-i described with reference to FIG. 2C, it is indicated by the wave algorithm data WAVEALG. By reading out a wave program, that is, one of the wave programs WAVEPRG1-j described in FIG.
Wave DSP (WAVE) specified by sequence number MLTNO
DSP (MLTNO)), that is, the wave DSP 161 or 1
63 is set in one of the program memories (PRGMEM) (step S12).

Similarly, in the voice data VOICE (VNO) specified by the voice number VNO, one of the effect programs indicated by the effect algorithm data NOTEFCTALG, ie, any of the effect programs NOTEFCTPRG1-k described with reference to FIG. And reads the effect DSP (EFCTDSP) specified by the sequence number MLTNO.
(MLTNO)), that is, the effect DSP 162 or 1
64 is set in one of the program memories (PRGMEM) (step S13).

Then, in the voice data VOICE (VNO) specified by the voice number VNO, the wave parameter data WA
VEPAR is read out, and a wave parameter buffer (WAVEPARBUF (MLTNO)) specified by the sequence number MLTNO,
That is, the data is transferred to one of the first and second wave parameter buffers (WAVEPARBUF1, 2) described in FIG. 2B (step S14).

Similarly, the effect parameter data NOTEFCTPAR is read out from the voice data VOICE (VNO) specified by the voice number VNO, and the effect parameter buffer (EFCTPARBUF (MCTNO) specified by the series number MLTNO is read.
LTNO)), that is, the first or second effect parameter buffer (EFCTPARBUF1,
After the transfer to any one of 2) (step S15), the main tone parameter setting processing ends.

FIG. 9 is a flowchart showing the detailed procedure of the sound generation instruction processing subroutine in step S6.

In the figure, first, a designated timbre corresponding to the generated key-on event x, ie, a sequence number, is detected and stored in a predetermined area MLTNOx of the RAM 7 (step S21). Hereinafter, the content stored in the area MLTNOx is referred to as a sequence number MLTNOx.

Next, in the wave DSP (WAVEDSP (MLTNOx)) designated by the sequence number MLTNOx, it is searched for an empty channel (step S22), and it is determined whether an empty channel is found (step S22).
23).

In step S23, when an empty channel is found, the process proceeds to step S24, and when it is not found, the process proceeds to step S29.

At step S24, sequence number MLTNO
Wave parameter buffer specified by x (WAVEPARB
UF (MLTNOx)) is changed according to the velocity (VELOCITYx) and the key code (KCx) corresponding to the key-on event x, and the result is stored in the key-on wave buffer KEYONWAVEBUF described in FIG. .

In a succeeding step S25, the contents of the key-on wave buffer KEYONWAVEBUF are stored in the sequence number MLT.
Wave DSP (WAVEDSP (MLTNO
x)) to the parameter register (PARREGx).

Similarly, the effect parameter buffer (EFCTPARBUF (MLT
NOx)) is changed according to the velocity (VELOCITYx) and key code (KCx) corresponding to the key-on event x, and the result is stored in the key-on effect buffer KEYONEFCTB.
The contents of the key-on effect buffer KEYONEFCTBUF are stored in the sequence number MLTNO (step S26).
Effect DSP specified by x (EFCTDSP (MLTNOx))
(PARREGx) (step S27).

Then, after an instruction to start sound generation is given to the channel x of each DSP (step S28), the main sound generation instruction processing ends.

In step S29, a channel to be truncated is determined. The truncation method includes, for example, a method of determining the channel with the most attenuation as a truncated channel, a method of determining the oldest key-on channel as a truncated channel, and the like.
There are various methods. However, the present invention is not limited to the truncation method, and any method may be employed.

In the following step S30, step S29
After the truncation process for truncating the channel determined in step S31 is performed, and until the answer is "YES" as a result of the determination in step S31, the process proceeds to step S31.
Proceed to 24.

This tone generation instruction processing is intended for a case where another key-on event does not occur after the occurrence of a key-on event and before the occurrence of the corresponding key-off event. Although this is not taken into account, it can be easily dealt with by repeatedly executing the main sounding instruction processing.

As described above, in the present embodiment,
A different effect program is set for each series, and by executing this effect program, a different effect is applied to each series, so that the user can apply the desired effect according to the tone color. Can be expanded.

Next, the effects DSP162,
FIG. 10 shows the ten types of effect giving processing executed in 164.
This will be specifically described with reference to FIGS.

FIG. 10 is a diagram for explaining a process of giving an effect (frequency characteristic) peculiar to an electromagnetic pickup of an electric piano. FIG. 10A shows the algorithm in hardware. (B) shows the structure around the electromagnetic pickup.

As shown in (b), the vibrator (Tine) and the pickup (Pickup) placed opposite to the vibrator (Tine) surround the electromagnetic pickup of the electric piano.
Mainly composed of a vibrator and a hammer (not shown)
The vibrator is vibrated by hitting with, and the pickup detects this vibration and converts it into an electric signal.

The algorithm (a) simulates the vibration of the vibrator detected by the electromagnetic pickup.

In (a), the input signal Input is multiplied by the value of the parameter Drive by the multiplier MLT1, and its amplitude is changed. Here, the parameter Drive is
A parameter indicating the magnitude of the amplitude of the vibrator. The larger this value is, the more the change in timbre with respect to the change in volume of the input signal Input is emphasized. This parameter Drive may be rephrased as changing the distance between the pickup and the vibrator.

The parameter Drive is underlined as shown in FIG. This underline indicates the input signal Input
(I.e., the basic tone waveform of a certain channel generated by the wave DSP 161 or 163; hereinafter, the same applies even if the drawing is changed). Specific information (a typical one is "pitch", so hereinafter, for convenience of explanation, " (It will be described by limiting to "pitch"). Specifically, for the parameter Drive, a shift amount for shifting the value of the parameter Drive in the positive or negative direction according to the pitch of the input signal Input is determined, and this shift amount is set to the value of the parameter Drive. It is controlled by adding. This control method is common to any parameters that are controlled according to the pitch. Of course, it is not necessary to limit to this control method, and any method may be adopted as long as the value of the parameter is changed according to the pitch.

The output of the multiplier MLT1 is branched into two,
One is supplied to an integrator Ineg, and this integrated output is supplied to an input terminal T0 of a selector SEL1 after unnecessary low frequency components are removed by a high-pass filter HPF1,
The other is supplied to the input terminal T1 of the selector SEL1.

The parameter Pickup_type is supplied to the select terminal of the selector SEL1, and the parameter Pickup_ty
When pe is “0” or “2”, the signal supplied to the input terminal T1 is output from the selector SEL1, and when the parameter Pickup_type is “1”, the signal supplied to the input terminal T0 is supplied from the selector SEL1. Is output.

Here, the parameter Pickup_type is 0 to
A parameter that takes any integer value of 2 and is "0"
Indicates that the selector SEL1 and the subsequent-stage selector 2 are both of a "normal" type in which the input signal on the input terminal T1 side is selected, and a "normal" type aiming at a shallower effect, and "1" treats the input signal Input as a velocity wave. And a displacement wave which is integrated by an integrator Ineg following the multiplier MLT1, and the output (corresponding to the magnetic flux density) of the non-linear table TBL is selected to select the output of the differentiator Diff. This indicates an "integrate" type in which signal processing is performed. In the mode of selecting "2", the selector SEL1 selects the input terminal T1 and the selector SEL2 selects the "differentiate" type in which the input terminal T0 is selected.

The output of the selector SEL1 is supplied to one input terminal of the adder ADD, and the other input terminal of the adder ADD is supplied with the parameter Position. Parameter Posi
“tion” is a parameter indicating the position of the transducer shifted from the center of the pickup, and the value of the parameter “Position” is controlled by the pitch.

FIG. 11 is a diagram showing an example of the characteristics of the shift amount parameter for shifting the parameter Position according to the pitch. The horizontal axis indicates the pitch, and the vertical axis indicates the magnitude of the shift amount parameter. I have. The parameter Ao described later
Since ut also performs the same control as the parameter Position, FIG. 11 also shows characteristics of a shift amount parameter for shifting the parameter Aout. For this reason, the names of the parameters are omitted on the vertical axis.

In the figure, the parameter low_break_poin
t indicates a breakpoint below “C3” for giving a change in the parameter Position due to the pitch, and a parameter high_break_point similarly indicates a breakpoint above “C3”.

Then, the parameter Position_high_kf is
Indicates the slope of the parameter Position above the pitch set by the parameter high_break_point. The value of the parameter Position is set to be larger as the slope is positive and higher, and the value of the parameter Position is set to be larger as the pitch is lower and lower. Similarly, the parameter Position_high_mid_kf is
Indicates the slope of the parameter Position between “C3” and the pitch set by the parameter high_break_point, and the parameter Position_low_mid_kf is “C3” and the parameter low_break_point.
Parameter Pos between the pitch set by break_point
The parameter Position_low_kf indicates the gradient of the parameter Position below the pitch set by the parameter low_break_point.

Returning to FIG. 10, the output of the adder ADD is input to a non-linear table TBL for simulating a phenomenon that a distortion component is added to a detection signal by an electromagnetic field generated by the electromagnetic pickup. An output signal from the adder ADD, that is, a signal obtained by adding an offset corresponding to the value of the parameter Position to the output signal from the selector SEL1, has a non-linear table T according to its signal level (amplitude).
BL is searched and a distortion component is added.

The output of the nonlinear table TBL is branched into two, one is supplied to the differentiator Diff, and the other is supplied to the high-pass filter HPF2. High pass filter H
A parameter Highpass indicating the frequency at which the low frequency of the pickup output is cut is supplied to PF2.

The output of the differentiator Diff is supplied to the selector SEL2.
And the output of the high-pass filter HPF2 is supplied to the input terminal T1 of the selector SEL2. The parameter Pickup_type is supplied to the select terminal of the selector SEL2. When the parameter Pickup_type is “2”, the signal supplied to the input terminal T0, that is, the signal differentiated by the differentiator Diff is output, and the parameter Pickup_type is set to “ Input terminal T1 when other than 2 "
, The high-pass filter HPF2
As a result, a signal from which low-frequency components have been cut is output.

The output of the selector SEL2 is supplied to a low-pass filter LPF that cuts a high-frequency component of the output signal. The low-pass filter LPF is for realizing (simulating) the characteristic of the inductance of the electromagnetic pickup, and has a parameter Cu indicating a cutoff frequency.
toff and parameters indicating the resonance characteristics of the electromagnetic pickup
Resonance is supplied. The parameter Cutoff is a parameter whose value is controlled according to the pitch.

The output of the low pass filter LPF is a multiplier M
The output signal Ou is multiplied by a parameter Aout which is supplied to the LT 2 and indicates the output level of the pickup which is supplied to the other.
tput is generated. Here, the parameter Aout is controlled by the pitch, but the control method is the same as the control method of the parameter Position described above, and the description thereof will be omitted.

Note that the present algorithm, as described above,
Software program (micro program) for effect imparting processing executed by effect DSPs 162 and 164
And is not implemented by hardware. However, FIG. 10 illustrates the present algorithm using hardware. This is because the explanation using hardware is intuitive and easy to understand.

FIGS. 12A and 12B are diagrams for explaining a process of giving a specific effect (frequency characteristic) in the relationship between the strings of the electric guitar and the electromagnetic pickup, and FIG.
FIG. 2 shows the algorithm in terms of hardware, and FIG. 2B shows the structure around the electromagnetic pickup.

Around the electromagnetic pickup of the electric guitar, as shown in (b), a string S stretched between a bridge B and a nut N, and an electromagnetic pickup provided at a predetermined position in the longitudinal direction of the string S ( Pickup), and the electromagnetic pickup detects a string vibration generated by the player playing the string S and converts it into an electric signal.

(A) simulates a period from when the player plays the string S to when the string is converted into an electric signal, that is, simulates a timbre change depending on a picking position and a pickup position and characteristics of an electromagnetic pickup. The algorithm of the program to be executed is shown. More specifically, according to the picking position and the pickup position, the input signal is converted into a string vibration, detected by an electromagnetic pickup, and output with a frequency characteristic unique to the pickup. In (a), an input signal Input is branched into two, one is supplied to one input terminal of an adder ADD1 via a delay unit Delay1, and the other is supplied to a multiplier M
The signal is supplied to the other input terminal of the adder ADD1 via LT1. The multiplier MLT1 is supplied with a parameter Picking_notch that gives a change in timbre according to the picking position, and the multiplier MLT1 receives the parameter Picking_notch.
The amplitude of the input signal Input is adjusted according to the value of
Output to DD1. The parameter Picking_notch can change the resonance peak characteristics of the comb filter formed by the feedforward path via the delay unit Delay1 and the multiplier MTL1 in various ways depending on the positive and negative polarities and values, thereby obtaining various changes in timbre. Can be When the value of the parameter Picking_notch is changed in a positive range, it corresponds to a simulation of the operation of changing the pluck position with a guitar or the like, and when the value is negative, the setting cannot occur in the actual string vibration phenomenon, New sounds and special effects can be produced.

The adder ADD2 is added to the delay unit Delay1.
Accordingly, when the position of the fret is changed, the value of the parameter Flet_control for simultaneously controlling the change in the picking position and the change in the position of the pickup and the value of the parameter Picking_position indicating the position of the picking are added and supplied. The value of the parameter Picking_position is, for example, “0”, just before the bridge B side, and “127”, indicating the center of the string S. Also, the parameter Picking_position
Is also a parameter controlled by the pitch, and its value is changed by a shift amount determined according to the pitch.

The output of the adder ADD1 is branched into two, one of which is added via a delay unit Delay2 to the adder ADD3.
, And the other is supplied to the other input terminal of the adder ADD3 via the multiplier MLT2. The multiplier MLT2 is supplied with a parameter Pickup_notch that gives a change in timbre depending on the position of the pickup. The multiplier MLT2 adjusts the amplitude of the input signal Input according to the value of the parameter Pickup_notch, and outputs the adjusted signal to the adder ADD3. The parameter Pickup_notch can change the resonance peak characteristic of the comb filter formed by the feedforward path via the delay unit Delay2 and the multiplier MTL2 in various ways depending on the positive and negative polarities and values, thereby obtaining various changes in timbre. Can be Parameter Pickup
When the value of _notch is changed in a positive range, it corresponds to a simulation when the position of the pickup with respect to the string S in the electric guitar is changed. On the other hand, when the value of _notch is set to a negative value, the setting cannot occur in reality, New sounds and special effects can be produced.

The delay unit Delay2 has an adder ADD4
Accordingly, the value of the parameter Flet_control and the value of the parameter Pickup_position indicating the position of the pickup are added and supplied. Parameter Pickup_position
Indicates that the value is, for example, "0", which is just before the bridge B side, and "127" indicates the center of the string S. The parameter Pickup_position is also a parameter controlled by the pitch, and its value is changed by a shift amount determined according to the pitch.

FIG. 13 is a diagram showing a string model and its equivalent conversion model. The delay units Delay 1 and Delay 2 shown in FIG.
Shows the theoretical basis that the string vibration generated in the string S can be simulated. In FIG. 13, (a) models the string vibration at the position of the electromagnetic pickup, that is, the string vibration generated when the string vibration generated at the string is transmitted to the position of the electromagnetic pickup, and (b) illustrates the string vibration.
3A shows an equivalently transformed model of FIG.

It is known that the string vibration generated in the string S when an excitation signal (vibration) is given is generally represented by a model shown in FIG. In (a), each block represents a delay device, and the delay amount of each delay device is determined according to the assumed effective length of the string as a vibrator, the position of picking, and the position of the electromagnetic pickup. Basically,
The delay amounts of the delay units corresponding to the upper and lower sides are equal. The total delay amount of the loop circuit, that is, the delay devices DL1, DL2, DM
1, DM2, DR1, DR2 total delay (dl1 + dl
2 + dm1 + dm2 + dr1 + dr2) is determined by the pitch of the generated musical tone, and the distribution of the delay amount of each delay unit is determined according to the position of picking and the position of the electromagnetic pickup.

In order to realize the vibration loss characteristics of the strings, an arbitrary signal attenuation element such as a filter may be appropriately inserted in the loop circuit. Further, the delay devices DR1 and DR1
In some cases, a filter or a phase inverter is installed between the two or the delay units DL1 and DL2. In this case, the installed element, that is, the filter or the position inverter is
What is necessary is just to consider that it is contained in any of the delay devices DR1, DR2, DL1, DL2.

When the model of (a) is equivalently transformed, (b)
Is obtained. In (b), the delay amount dm of the delay unit DM is equal to the delay amount dm of each of the delay units DM1 and DM2.
1, dm2 (dm = dm1 + dm2), and similarly, the delay amount dr of the delay device DR and the delay device DL
And the delay amount dx of the delay unit DX correspond as follows.

Dr = dr1 + dr2 dl = dl1 + dl2 dx = dt (= dr + dl + dm) -dl = dr + dm By equivalently converting (or simplifying) the model of (b) again, the delay units Delay1, Delay2 and adder of FIG. A circuit composed of 1 and 3 is obtained. That is,
With the circuit of FIG. 12, it is possible to simulate the string vibration generated in the string S. Of course, the configuration shown in FIG. 13A or 13B may be used as it is for the string model portion in FIG.

Returning to FIG. 12, the output of the adder ADD3 is
The integrated output is supplied to an integrator Integra, and the integrated output is input to a non-linear table TBL for simulating a phenomenon that a distortion component is added to a detection signal by an electromagnetic field generated by the electromagnetic pickup. Non-linear table TBL
Includes a parameter Di indicating the distance between the pickup and the string S.
stance is supplied. The value of the parameter Distance is offset (added) to, for example, the integrated output from the integrator Integra, and the nonlinear table TBL is searched based on the offset result, and a signal to which a predetermined distortion component is added is output. The parameter Distance is, for example,
The smaller (closer) the value is, the higher the output level is, which acts to emphasize the distortion of the electromagnetic pickup.

The output of the non-linear table TBL is
After being supplied to iff and differentiated, the signal is supplied to an emphasizer (operating substantially equivalent to an equalizer) Emp that emphasizes a predetermined frequency component. Emphasizer Emp
The parameter Pickup indicates the type of electromagnetic pickup
_type is supplied, and the emphasizer Emp emphasizes the frequency component of the input signal according to the value of the parameter Pickup_type and according to the pickup. Parameters
Pickup_type takes, for example, an integer value of either 0 or 1. When “0”, the frequency component is emphasized so as to have a high-pitched sound peculiar to a single-coil pickup, and when “1”, the hum is used. The frequency component is emphasized so as to make the mellow sound peculiar to the backing pickup.

The output of the emphasizer Emp is supplied to a high-pass filter HPF, and the high-pass filter HPF is supplied with a parameter Highpass indicating a frequency at which a low band of the pickup output is cut. This parameter High
The pass is also controlled by the pitch.

The output of the high-pass filter HPF is branched into two, one of which is directly supplied to one input terminal of a mixer MIX which can change the mixing amount, and the other of which is supplied via a low-pass filter LPF to the mixer MIX. It is supplied to the other input. The low-pass filter LPF is supplied with a parameter Cutoff indicating a cut-off frequency of the filter that realizes the characteristic due to the inductance of the electromagnetic pickup and a parameter Resonance indicating the resonance characteristic of the electromagnetic pickup.
A signal from which high-frequency components have been cut in accordance with toff and Resonance is supplied to the other input terminal of the mixer MIX.

The mixer MIX has a ratio of bypassing the characteristics of the electromagnetic pickup, that is, a low-pass filter LP.
A parameter Filter_bypass indicating a mixing ratio between the signal whose high-frequency component has been cut by F and the signal bypassing the low-pass filter LPF is supplied to the mixer MIX.
Mixes the output signal from the low-pass filter LPF and the signal directly output from the high-pass filter HPF according to the value of this parameter Filter_bypass, and outputs
Output to the outside as ut.

FIG. 14 is a diagram for explaining a process of giving an effect of changing an input signal into a water sound, and shows the algorithm in terms of hardware. In this algorithm, a sample and hold LFO (Lo
w Frequency Oscilator) is applied to the filter modulation to create the feeling of flowing water, and the pitch is given by passing through a resonant string according to the pitch.

In the figure, an input signal Input is branched into two, one of which is supplied to a band-pass filter BRF,
The other is supplied to a multiplier MLT1.

The adder A is added to the band-pass filter BRF.
The output of DD1 and a parameter Resonance indicating the resonance of the filter BPF are supplied.

One input terminal of the adder ADD1 has a parameter Cu indicating the center frequency of the filter modulation.
toff is supplied, and the output of the multiplier MLT2 is supplied to the other input terminal. In the multiplier MLT2, the noise generated by the noise generator Noise is sampled / held by the sample / hold circuit S / H according to the signal generated by the ultra-low frequency oscillator LFO, and the high-frequency component is output by the low-pass filter LPF. Is removed, and this signal is multiplied by a value of a parameter Depth that indicates the depth of the filter modulation that has been input to another signal. Then, the circuit from the noise generator Noise to the multiplier MLT2 uses the sample & hold LFO.
Is configured.

The very low frequency oscillator LFO has a sample &
A parameter Speed indicating the speed of the hold LFO is supplied. The larger the value of the parameter Speed, that is, the faster it sounds, the faster the flow of water sounds.
Furthermore, a parameter Smoothness indicating the smoothness of the sample & hold LFO is supplied to the low-pass filter LPF. The parameter Smoothness sounds like the viscosity of the water changes according to its set value.

The output of the band-pass filter BRF is supplied to one input terminal of an adder ADD2.
Is supplied with the output of the multiplier MLT3,
The result of the addition is supplied to the low-pass filter LPF2.

The low-pass filter LPF2 is supplied with a parameter High_dump indicating the amount of high-frequency attenuation in the resonance string. The parameter High_dump specifically corresponds to the material of the string.

The output of the low-pass filter LPF2 is branched into two, one of which is supplied to the delay unit Delay, and the other is supplied to the adder ADD3. The delay device Delay
Parameter Pitch_co that sets the pitch of the resonant string in semitone units
arse and a parameter Pitch_fine for finely adjusting the pitch of the resonance string in cent units are supplied.

The output of the delay unit Delay is branched into two, one of which is an output signal Output1 to the outside, and the other is supplied to the multiplier MLT3. The multiplier MLT3 is also supplied with a parameter FB indicating the resonance strength of the resonance string, and the amplitude of the output signal from the delay unit Delay is changed by the parameter FB, that is, the feedback amount is adjusted. Here, as the value of the parameter FB increases, the sense of pitch becomes clearer. However, if the value is too large, the atmosphere of the water sound disappears.

The output of the envelope generator EG is supplied to the other input terminal of the adder ADD3. The envelope generator EG is supplied with a parameter Release indicating the decay time of the output signal after the note-off. Then, an envelope signal is generated according to this parameter Release.

The output of the adder ADD3 is supplied to the high-pass filter HPF. The high-pass filter HPF is supplied with a parameter Highpass indicating the cutoff frequency of the high-pass filter HPF applied to the water sound output. Is supplied to the multiplier MLT4.

To the multiplier MLT4, a parameter Wet indicating the output level of the water sound is also supplied.
After the output level of the water sound is adjusted according to the value of et, it is supplied to one input terminal of the adder ADD4.

The output of the multiplier MLT1 is supplied to the other input terminal of the adder ADD4. The multiplier MLT1 includes:
A parameter Dry indicating the output level of the input signal Input is supplied. After the output level of the input signal Input is adjusted according to the value of the parameter Dry, the parameter is supplied to the adder ADD1.

The adder ADD1 adds the input signal Input and a signal indicating a water sound, that is, the output signal from the multiplier MLT4, generates an output signal Output2, and outputs it to the outside.

FIG. 15 shows the PWM of the analog synthesizer.
FIG. 9 is a diagram for explaining a process of giving an effect similar to the (Pulse Width Modulation) effect, and shows the algorithm in terms of hardware.

The present algorithm is configured based on the following principle.

That is, when the difference between the phase-shifted sawtooth waveform and the original waveform is obtained, a pulse waveform is obtained. Using this principle, the difference between the input waveform passed through the delay unit and the input waveform is calculated, and the readout position (phase) of the delay unit is modulated by an ultra-low frequency oscillator or envelope generator, etc. A PWM waveform is obtained. Eventually, it works as a comb filter, so that any input will attenuate certain harmonics. If the delay is exactly half the period of the input signal, even harmonics will be canceled out and only odd harmonics will be present. By obtaining such comb-shaped frequency characteristics, the analog synthesizer VC
Frequency characteristics similar to PWM of O (Voltage Controlled Oscillator) can be simulated. As a result, a chorus feeling having a “taste” like PWM can be given to any waveform input.

In FIG. 15, an input signal Input is branched into two, one of which is a delay device Delay whose delay amount is determined according to various parameters described later and an input value “−”.
1 "is supplied to one input terminal of a mixer MIX which can change the mixing amount via a multiplier MLT1 which inverts an input signal (that is, an output signal from the delay unit Delay),
The other is supplied directly to the other input of the mixer MIX.
The mixer MIX is supplied with a parameter Balance indicating a mixing amount of the input signal Input and the delayed signal, and a parameter B
alance is "0" and only input signal Input is output signal Output
And the difference between the input signal Input and the delayed signal at "32", the signal with the normal PWM effect added is output as the output signal Output, and the sum at "-32" gives the chorus effect The output signal is output as an output signal Output.

The output of the adder ADD having three input terminals is supplied to the delay unit Delay.
Delays the input signal Input by an amount of delay determined according to the output value of the adder ADD.

The three input terminals of the adder ADD are supplied with a parameter Pulse_width for setting a pulse width, a parameter Pitch_coarse for setting a delay length in semitone units, and an output of a multiplier MLT2 having three input terminals. The parameter Pulse_width is, for example, “64” and a pulse width of 50.
%, And the parameter Pitch_coarse is
For example, when "0" is set, a delay length corresponding to C3 = 261.63 Hz is set.
Is set to the basic pitch.

The three inputs of the multiplier MLT2 receive the output of the envelope generator EG, the parameter PWM_depth indicating the depth of the pulse width modulation, and the output of the very low frequency oscillator LFO whose amplitude has been adjusted via the multiplier MLT3. Supplied.

A parameter EG_mode indicating the operation mode, a parameter EG_shape indicating the shape (depth) of the envelope, and a parameter Time indicating the arrival time of the envelope are supplied to the envelope generator EG.
The parameter EG_mode takes an integer value of, for example, 0 to 2. When "0", the envelope of the decay portion of the musical tone waveform is generated by the envelope generator EG, and when "1", the envelope of the attack portion is generated. , "2", a fade-in envelope is generated. When the parameter EG_shape is “0”, no envelope is generated (the value remains at the maximum value),
The depth is 50% for "32" and 100% for "64". However, when fade-in is specified, “3
The delay unit Delay operates at 2 "or more. Parameter Tim
e is a parameter controlled by the pitch, and the arrival time of the envelope is changed by the pitch.

The very low frequency oscillator LFO includes a parameter LFO_mode indicating its operation mode, a parameter Speed indicating its frequency, a parameter Wave indicating the type of output waveform, and a very low frequency oscillator LFO reset by note-on.
Is supplied, and the very low frequency oscillator LFO generates an output waveform corresponding to these various parameters. Here, the parameter LFO_mode takes an integer value of, for example, any one of 0 to 2, “0” indicates a mode in which the ultra-low frequency oscillator LFO is used in common for all notes, and “1” indicates a mode. The low-frequency oscillator LFO is used for each note to indicate a mode in which reset is performed at the time of note-on, and “2” indicates the mode in which the ultra-low-frequency oscillator LFO is used for each note and the phase at the time of note-on changes randomly. ing. The parameter Wave takes an integer value of either 0 or 1, for example, and “0” is the very low frequency oscillator L
Indicates the mode in which a triangular wave is output from the FO, and “1” indicates
The mode in which a sine wave is output from the very low frequency oscillator LFO is shown. The parameter Phase is, for example, 0 °, 90 °
°, 180 °, and 270 °.

The multiplier MLT3 has a parameter De indicating the pulse width modulation depth in the very low frequency oscillator LFO.
pth is supplied, and the output signal from the very low frequency oscillator LFO is supplied to the multiplier MLT2 after the amplitude is adjusted by the parameter Depth. The parameter Depth is
For example, when "0", the output from the very low frequency oscillator LFO becomes invalid and only the output from the envelope generator EG becomes valid, and when "32", the amplitude of the output from the very low frequency oscillator LFO increases from zero. It becomes possible to swing up to the amplitude of the envelope of the envelope generator EG, and when it is "64", the very low frequency oscillator LFO
Are positive and negative output signals whose absolute value of the amplitude of the output from the envelope generator EG swings up to the amplitude of the output from the envelope generator EG.

FIG. 16 shows a flanger (flange) which is an effector for imparting a sound like a rising and falling sound of a jet.
FIG. 9 is a diagram for explaining a process of giving the effect generated by r), and shows the algorithm in hardware.

The present algorithm is based on the following principle.

That is, a dip (trough) is generated at a plurality of frequencies by mixing an input signal delayed through a delay unit with the input signal, and a plurality of dips are generated by feeding back the output from the delay unit. Generate peaks (mountains). By modulating the delay length with an envelope generator or an ultra-low frequency oscillator, it operates as a special filter that changes the frequency of a plurality of dips and peaks. A special effect can be obtained by accurately associating the delay length with the pitch frequency.

In FIG. 16, an input signal Input is branched into two, one is supplied to one input terminal of an adder ADD1, and the other is supplied to one input terminal of a multiplier MLT1. The other input terminal of the multiplier MLT1 has an input signal Inpu
The parameter Dry indicating the output level of t is supplied, and the input signal Input is supplied to one input terminal of the adder ADD2 after the amplitude is adjusted according to the value of the parameter Dry.

At the other input terminal of the adder ADD1, the output of the delay unit Delay whose delay amount is determined according to various parameters described later indicates the feedback amount (the feedback amount of the flanger) by the multiplier MLT2. Supplied after the amplitude is adjusted according to the value of the parameter Feedback. The larger the absolute value of the parameter Feedback, the more the input signal Inpu
The peak at the frequency of t is emphasized.

The output of the adder ADD1 is a delay unit Delay
, And the output of the delay unit Delay is branched into two, one of which is supplied to the multiplier MLT2, and the other is supplied to one input terminal of the multiplier MLT3. Multiplier MLT
A parameter Wet indicating the output level of the signal to which the flanger effect has been applied is supplied to the other input terminal of the multiplier 3. Parameter We
The amplitude is adjusted by the value of t and the other input of the adder ADD2 is simply supplied.

The output of the adder ADD2 is the output of the multiplier MLT4
And the other input terminal of the multiplier MLT4 is provided with an envelope generator EG1 that outputs an envelope whose decay time is controlled in accordance with a parameter Release indicating the decay time of the output signal after note-off. Output is provided. The multiplier MLT4 adjusts the amplitude of the output from the adder ADD2 according to the value of the envelope from the envelope generator EG1, and outputs it as an output signal Output to the outside.

The delay unit Delay has an adder ADD3 for adding the envelope output from the envelope generator EG2 and the output signal from the very low frequency oscillator LFO.
Is supplied, and the delay unit Delay delays the output signal from the adder ADD1 by a delay amount according to the addition result.

The envelope generator EG2 has a parameter Attac indicating the time of the attack portion of the envelope.
k, parameter D indicating the time of the decay part of the envelope
ecay and a parameter Sustain indicating the envelope reaching level are supplied, and the envelope generator EG
2 generates an envelope based on these parameters and supplies the envelope to the multiplier MLT5. The multiplier MLT5 is supplied with a parameter EG_depth for adjusting the output level, and the multiplier MLT5 calculates the amplitude of the envelope from the envelope generator EG2 and the parameter EG_dep.
The value adjusted by the value of th is supplied to the adder ADD3.

Here, the parameter Attack is a parameter controlled by the pitch, and the parameter Decay is also
This is a parameter controlled by the pitch.

The very-low-frequency oscillator LFO is supplied with a parameter Speed indicating the frequency of the output signal, and the very-low-frequency oscillator LFO generates a signal having a frequency corresponding to the value of the parameter Speed, and outputs the signal to the multiplier MLT6. To supply. The multiplier MLT6 is supplied with a parameter LFO_depth for adjusting the output level, and the multiplier MLT6 calculates the amplitude of the output from the very low frequency oscillator LFO and the parameter LFO_dept.
The value adjusted by the value of h is supplied to the adder ADD3.

FIG. 17 is a diagram for explaining a process of giving an effect generated by a phaser (Phaser), which is an effector for imparting a sense of rotation, spread and depth to a tone signal. It is shown.

The present algorithm is based on the following principle.

That is, by mixing the input signal with an ancestor delay through an all-pass filter and the input signal, dips (valleys) are generated at a plurality of frequencies and the output from the all-pass filter is fed back. Therefore, a plurality of peaks (mountains) are generated. By modulating this all-pass coefficient with an envelope generator or an ultra-low frequency oscillator, it operates as a special filter in which the frequencies of a plurality of dips and peaks change. A special effect can be obtained by accurately associating the phase change with the pitch frequency.

As can be seen from FIG. 17, the effect processing differs from the effect processing of FIG. 16 in that an all-pass filter APF is used instead of the delay unit Delay, and the type of parameters supplied to the envelope generator EG2 is different. Only the difference is described, and only this difference will be described.

The all-pass filter APF receives the input signal In.
Used to add phase delay to put.

The envelope generator EG2 is supplied with a parameter EG_mode indicating its operation mode and a parameter Time indicating the arrival time of the envelope.
The parameter EG_mode takes, for example, an integer value of either 0 or 1. When "0", the envelope generator EG generates a decay type, that is, an attenuating type envelope, and when "1", an attack type, that is, an increasing type. This type of envelope is generated.

FIG. 18 is a diagram for explaining a process of giving an effect of adding a harmonic component to an input signal, and shows the algorithm in hardware.

This effect can be realized by inputting the input signal to the delay unit and modulating the read position (phase) by the input signal, thereby giving a distortion to the waveform.

In FIG. 18, an input signal Input is branched into three, one is supplied to one input terminal of a multiplier MLT1 and then supplied to one input terminal of an adder ADD1. The signal is supplied to the other input terminal of the adder ADD1 via the delay unit Delay and the multiplier MLT2, and the other one is supplied to a low-pass filter LPF that cuts a high-frequency component of the input signal Input. The adder ADD1 adds the two inputs and outputs the result as an output signal Output.

The multiplier MLT1 is supplied with a parameter Dry indicating the output level of the input signal Input.
ut is adjusted in amplitude according to the value of this parameter Dry. Further, a parameter Wet indicating the output level of the modulated signal is supplied to the multiplier MLT2.
2 adjusts the output from the delay unit Delay, that is, the amplitude of the modulated signal by the value of the parameter Wet.

To the delay unit Delay, the parameter Pitch_coarse for setting the delay length in semitone units and the output of the adder ADD2 having three input terminals are supplied. The parameter Pitch_coarse is, for example, “0” and C3 = 261.6.
The delay length is set so as to correspond to 3 Hz. However, if the modulation is to be deeper, the value is reduced. If it is set to “−12” from the basic pitch, the delay length becomes twice, and if it is set to “−24”, the delay length becomes four times.

At the three input terminals of the adder ADD2, the read position (phase) of the delay unit Delay, which is the center of modulation,
, The parameter Polarity indicating the direction of modulation, and the output of the multiplier MLT2. By changing the value of the parameter Phase, the phase relationship between the input of the delay unit Delay and the read position changes, and the timbre of the input signal Input changes. The parameter Polarity is, for example, a parameter that takes an integer value of either 0 or 1. When it is “0”, the delay device D is configured so that the delay amount increases as the input signal Input increases.
The delay is controlled, and when "1", the delay unit Delay is controlled such that the delay amount decreases as the input signal Input increases.

The output of the low-pass filter LPF is supplied to a multiplier MLT2, and an adder ADD3
Is supplied. The low-pass filter LPF is supplied with a parameter Drive_lowpass indicating a frequency at which a high-frequency component of the input signal Input used for modulation is cut. The parameter Drive_lowpass is a parameter controlled by the pitch.

The parameters Driv set in this way are
In response to e_lowpass, the low-pass filter LPF supplies the input signal Input from which the high-frequency component has been removed to the multiplier MLT2.

The adder ADD3 is supplied with the parameter Drive indicating the modulation depth and the envelope output from the envelope generator EG whose amplitude has been adjusted via the multiplier MLT3. The parameter Drive is also a parameter controlled by the pitch.

The envelope generator EG is supplied with a parameter EG_mode indicating its operation mode and a parameter Time indicating the arrival time of the envelope. The parameter EG_mode takes, for example, an integer value of either 0 or 1. When "0", the envelope generator EG generates a decay type, that is, an attenuating type envelope, and when "1", an attack type, that is, an increasing type. This type of envelope is generated. The envelope generator EG generates an envelope according to these parameters and outputs the generated envelope to the multiplier MLT3.

The multiplier MLT3 is supplied with a parameter EG_depth for adjusting the output level, and the multiplier MLT3
Adjusts the amplitude of the envelope from the envelope generator EG according to the value of the parameter EG_depth and supplies the adjusted amplitude to the adder ADD3. The adder ADD3 adds the amplitude-adjusted envelope and the value of the parameter Drive, and supplies the result to the multiplier MLT2. Multiplier MLT
Calculates the amplitude of the input signal Input, from which the high-frequency component output from the low-pass filter LPF has been cut, by the adder ADD3
Is adjusted according to the output value from the adder and supplied to the adder ADD2.

FIG. 19 is a diagram for explaining a process of giving an effect by a non-carrier (carrier frequency is zero) FM operator using an input signal as a modulator, and shows the algorithm in hardware. It is.

The present algorithm is based on the following principle.

That is, an input signal is processed to obtain a modulator waveform, and a sine wave is read out with this. When the modulation is deep, the phase change to be read is large, so that a high frequency component is emphasized, and when the modulation is shallow, the low frequency component is mainly. Further, the waveform read out can be changed from a sine wave to a sawtooth wave having many overtones by the feedback FM. In this way, from the so-called FM sound,
You can get even the sound of an analog synthesizer.

In FIG. 19, an input signal Input is branched into two, one is supplied to a multiplier (amplifier) MLT1, and the other is supplied to a multiplier MLT2. Multiplier MLT
1 is a parameter D indicating the output level of the input signal Input.
ry is supplied, and the multiplier MLT1 outputs the parameter Dry
The amplitude of the input signal Input is adjusted according to the value of
It is supplied to one input terminal of DD1.

The multiplier MLT2 is supplied with a parameter Pre_gain indicating a gain for amplifying the input signal Input used for modulation. The multiplier MLT2 amplifies the amplitude of the input signal Input according to the value of the parameter Pre_gain, and outputs a low-pass signal. It is supplied to the filter LPF1.

The low-pass filter LPF has a parameter Pre_lowp indicating a frequency at which a high-frequency component of the modulated signal is cut.
ass is supplied, and the low-pass filter LPF removes a high-frequency component of the output signal from the multiplier MLT2 according to the value of the parameter Pre_lowpass, and removes one input terminal of the multiplier MLT3 having three input terminals. To supply.

The multiplier MLT3 is supplied with the parameter Drive indicating the modulation depth and the output of the multiplier MLT4. The parameter Drive is a parameter controlled by the pitch.

A parameter EG_depth for adjusting the output level is supplied to the multiplier MLT4.
Adjusts the amplitude of the envelope from the envelope generator EG1 according to the value of the parameter EG_depth, and supplies the adjusted value to the multiplier MLT3.

The output of the envelope generator EG1 is branched into two, one of which is supplied to the multiplier MLT4 and the other is supplied to the multiplier MLT5. Multiplier MLT
5 is supplied with a parameter Edge_EG_depth indicating a modulation amount for modulating the FM feedback amount with the envelope output from the envelope generator EG1.
LT5 sets the amplitude of the input envelope as a parameter.
It is adjusted according to the value of Edge_EG_depth and is supplied to one input terminal of the adder ADD2.

The other input terminal of the adder ADD2 has FM
The parameter Bias indicating the feedback amount of
The adder ADD2 adds the amplitude-adjusted envelope and the value of the parameter Bias, and supplies the result as a parameter Edge_EG_depth to one input terminal of the multiplier MLT6.

The output of the multiplier MLT3 is supplied to a low-pass filter LPF2.
Is supplied with a parameter Ceiling indicating the upper limit of the frequency of the harmonic component to be emphasized. In accordance with the value of the parameter Ceiling, the low-pass filter LPF2 controls the multiplier MLT3
And removes the high-frequency component of the output signal from the adder ADD3 and supplies it to one input terminal of the adder ADD3. A parameter Overtone for controlling a harmonic component is supplied to the other input terminal of the adder ADD3. As the parameter Overtone has a larger set value, the output signal Output has an even number of overtones.

The output of the adder ADD3 is
And the other input terminal of the adder ADD4 is supplied with the output of the multiplier MLT6. The adder ADD4 adds these input values and supplies the sum to the sine wave table TBL. In the sine wave table TBL,
A sine wave is read out according to the input signal, and its output is fed back via the multiplier MLT6 and supplied to the high-pass filter HPF.

The high-pass filter HPF has a parameter Hi indicating the frequency at which the low-frequency component of the modulated signal is cut.
ghpass is supplied, and the high-pass filter HPF cuts a low-frequency component of the output signal from the sine wave table TBL according to the parameter Highpass, and outputs a low-pass filter LPF.
Supply to PF3. The parameter Highpass is a parameter controlled by the pitch.

The low-pass filter LPF3 has a parameter indicating a frequency at which a high-frequency component of the modulated signal is cut.
Lowpass is supplied, and the lowpass filter LPF3 is controlled by the highpass filter HPF in accordance with the parameter Lowpass.
Cuts the high frequency component of the output signal from the adder ADD5
To one of the input terminals. The parameter Lowpass is also a parameter controlled by the pitch.

An envelope from the envelope generator EG2 is supplied to the other input terminal of the adder ADD5, and the adder ADD5 adds these values and supplies the result to the multiplier MLT7.

The multiplier MLT7 is supplied with a parameter Wet indicating the output level of the modulated signal, and
T7 is an adder ADD according to the value of this parameter Wet.
5 and adjusts the amplitude of the output signal from the adder ADD1.
Is supplied to the other input terminal.

The adder ADD1 adds the output from the multiplier MLT1 and the output from the adder ADD1, and outputs the result as an output signal Output.

FIG. 20 is a diagram for explaining a process for providing an effect when an input signal (modulated wave) is subjected to AM modulation by a signal of an internal oscillator, and the algorithm is shown in hardware. Things.

In the present algorithm, when the degree of modulation is maximum (for example, 200%), multiplication is performed, which is equivalent to a so-called ring modulator, and generates a frequency component of the sum and difference of the modulated wave and the modulated wave. As the modulation degree decreases, the frequency component of the modulated wave becomes dominant, and at the modulation degree of 0%, only the modulated wave is present. In this algorithm, the modulating wave is a sine wave, but AM
By preparing two main and sub modulators, more complex harmonics can be added.

In FIG. 20, an input signal Input is supplied to one input terminal of a multiplier MLT1 having three input terminals, and the other two input terminals of the multiplier MLT1 receive the output of the main oscillator MainOSC and The output of the adder ADD1 is provided.

The output of the multiplier MLT1 is supplied to one input of a multiplier MLT2 also having three inputs, and the other two inputs of the multiplier MLT2 are connected to the output of the sub-oscillator SubOSC and the output of the sub-oscillator SubOSC. The output of the adder ADD2 is provided. The multiplier MLT2 multiplies all signals supplied to the three input terminals and outputs the result as an output signal Output to the outside.

The main oscillator MainOSC has a parameter Pitch_coarse for setting the pitch of the modulated wave on the main oscillator MainOSC side in semitone units, and a main oscillator MaiOSC.
A parameter Pitch_fine for finely adjusting the pitch of the modulated wave on the nOSC side in cent units, a main oscillator MainOS
Parameter Main indicating frequency offset of modulated wave on C side
_frequency_coarse, this parameter Main_frequency_co
Parameter Main_frequency_fine for fine adjustment of arse,
And the output of a multiplier MLT3.

A multiplier ML is provided to the sub oscillator SubOSC.
Output from T4, parameter Sub_frequency_ indicating the frequency offset of the modulated wave on the main oscillator MainOSC side
coarse, parameter Sub_frequency_fine for fine-tuning this parameter Sub_frequency_coarse, and multiplier M
The output of LT3 is provided. The parameters Pitch_coarse and Pitch_fine are connected to one input terminal of the multiplier MLT4.
Is supplied to the other input terminal, and the sub-oscillator SubOS
The pitch of the modulated wave on the C side is determined by the main oscillator MainOSC.
Parameter Sub_pitc to be set as a ratio to the side pitch
h is supplied. The multiplier MLT4 has a parameter Pitch_c
The values of oarse and Pitch_fine are changed at the ratio of the parameter Sub_pitch and supplied to the sub oscillator SubOSC.

The multiplier MLT3 is supplied with an envelope output from the envelope generator EG and a parameter PEG_depth for adjusting the output level.
The multiplier MLT3 adjusts the amplitude of the envelope from the envelope generator EG according to the value of the parameter PEG_depth, and adjusts the main and sub oscillators MainOSC,
Supply to SubOSC.

The envelope output from the envelope generator EG is also supplied to multipliers MLT5 and MLT6.
Further, a parameter Main_EG_depth for adjusting the output level is supplied to the multiplier MLT5.
Adjusts the amplitude of the envelope from the envelope generator EG according to the value of the parameter Main_EG_depth, and supplies the adjusted signal to one input terminal of the adder ADD1. Further, a parameter Sub_EG_depth for adjusting the output level is supplied to the multiplier MLT6.
The amplitude of the envelope from the envelope generator EG is adjusted by the value of the parameter Sub_EG_depth and supplied to one input terminal of the adder ADD2.

The envelope generator EG is supplied with a parameter EG_mode indicating the operation mode and a parameter Time indicating the arrival time of the envelope.
Generate an envelope corresponding to e and Time. The parameters EG_mode and Time are the same as the parameters EG_mode and Time described with reference to FIG.

The other input terminal of the adder ADD1 is supplied with a parameter Main_Mod_depth indicating the modulation depth on the main oscillator MainOSC side, and the other input terminal of the adder ADD2 is supplied with the modulation depth on the sub-oscillator SubOSC side. Is supplied as a parameter Sub_Mod_depth. Adder A
The DD1 adds the value of the parameter Main_Mod_depth to the envelope adjusted by the value of the parameter Main_EG_depth, and supplies it to the multiplier MLT1. The adder ADD2 adds the value of the parameter Sub_Mod_depth to the envelope adjusted by the value of the parameter Sub_EG_depth, and supplies the resultant to the multiplier MLT2.

FIG. 21 is a diagram for explaining a process of giving an effect when an input signal is passed through a low boost filter and then output through an overdrive circuit. The algorithm is shown in hardware. Things.

In the present algorithm, since the amount of low boost can be controlled by EG, not only the sound can be made thicker, but also the effect of a compressor can be obtained by emphasizing the attack portion. Since overdrive also works for each note, an effect different from that of a normal effector can be obtained.

In FIG. 21, the input signal Input is supplied to one input terminal of a multiplier MLT1, and the other input terminal of the multiplier MLT1 is supplied with a parameter Ain indicating the level of the input signal Input. Is the parameter A
The amplitude of the input signal Input is adjusted according to in and supplied to the low boost filter LBF.

The output of the adder ADD is supplied to the low boost filter LBF. A parameter Drive indicating the amount of low boost and the envelope output from the envelope generator EG are supplied to the multiplier MLT2 at two input terminals of the adder ADD. Are output, and the adder ADD adds these outputs to determine the final amount of low boost, and the low boost filter LBF
To supply.

The envelope generator EG has a parameter Attack indicating the time of the attack portion of the envelope,
Parameter Deca indicating the time of the decay part of the envelope
y and the parameter S indicating the level of the envelope reached
ustain is supplied and the envelope generator EG
An envelope is generated based on these parameters and supplied to the multiplier MLT2. The multiplier MLT2 includes:
A parameter Depth for adjusting the output level is supplied, and the multiplier MLT2 includes an envelope generator EG.
Is adjusted by the value of the parameter Depth and supplied to the adder ADD.

The output from the low boost filter LBF is supplied to one input terminal of a multiplier MLT2, and the other input terminal of the multiplier MLT2 is supplied with a parameter Over_drive indicating an overdrive amount.
After adjusting the amplitude of the output from the low boost filter LBF according to the value of the parameter Over_drive, when the output value exceeds a predetermined upper limit, the output value approaches the upper limit,
When the output value falls below a predetermined lower limit value, the output value is supplied to a saturation table TBL composed of table data having such characteristics as to approach the lower limit value.

The saturation table TBL searches for table data according to the input signal, and supplies the search result to the high-pass filter HPF. High pass filter HPF
Is supplied with a parameter Highpass indicating a frequency at which a low-frequency component of the output signal is cut off, and a high-pass filter HPF
Is a saturation table T according to this parameter Highpass.
The low-frequency component of the output signal from BL is cut off,
Supply to T3. The parameter Highpass is a parameter controlled by the pitch.

The multiplier MLT3 is supplied with a parameter Aout indicating an output level. The multiplier MLT3 adjusts the amplitude of the output from the high-pass filter HPF according to the value of the parameter Aout, and outputs it as an output signal Output. Output.

As described above, in the present embodiment, the values of the parameters (some parameters in the present embodiment) used in each effect algorithm are converted into the input signals (the basic tone signals generated by the wave DSPs 161 and 163). Since the control is performed in accordance with the pitch of the sound, that is, for each note, the range of the effect to be applied can be further expanded.

When the pitch of a musical tone to which an effect is to be applied changes every moment during sound production, the effect to be applied may be controlled in real time based on the frequency data of the musical tone.

The signal processing circuits shown in FIG. 10 and FIGS. 12 to 21 are circuit representations of signal processing algorithms performed by the DSP. In addition to using a DSP, a general-purpose microprocessor (for example, the CPU
It is also possible to process in 5). Of course, these may be implemented as hardware as shown.

A storage medium storing a software program for realizing the functions of the above-described embodiments is stored in a storage medium.
It goes without saying that the object of the present invention can also be achieved by supplying the program to a system or an apparatus and causing a computer (or CPU 5 or MPU) of the system or the apparatus to read and execute a program stored in a storage medium.

In this case, the program itself read from the storage medium realizes the novel function of the present invention,
The storage medium storing the program constitutes the present invention.

Examples of the storage medium for supplying the program include a hard disk of the HDD 11 and
D-ROM 21, MO, MD, floppy disk 2
0, a CD-R (CD-Recordable), a magnetic tape, a nonvolatile memory card, a ROM, and the like can be used.
In addition, other MIDI devices 100 and communication network 10
1, the program may be supplied from the server computer 102.

When the computer executes the readout program, not only the functions of the above-described embodiments are realized, but also the OS and the like running on the computer are actually executed based on the instructions of the program. It goes without saying that a part or all of the above-described processing is performed, and the functions of the above-described embodiments are realized by the processing.

Further, after the program read from the storage medium is written into the memory provided on the function expansion board inserted into the computer or the function expansion unit connected to the computer, the program is executed based on the instructions of the program. It goes without saying that the CPU 5 or the like provided in the expansion board or the function expansion unit performs part or all of the actual processing, and the processing realizes the functions of the above-described embodiments.

[0210]

As described above, according to the present invention,
A basic tone waveform is generated in a plurality of channels in accordance with the tone and pitch designated by the tone designation means, and an effect algorithm program corresponding to the designated tone is executed, and the effect is generated in the basic tone waveform in which the tone is generated. Is given independently for each channel, and the characteristics of the effect to be given are independently controlled for each channel according to the pitch of the basic tone waveform to which the effect of each channel is applied. This produces an effect that can expand the range of the effect.

[Brief description of the drawings]

FIG. 1 is a block diagram showing a schematic configuration of a musical sound synthesizer according to an embodiment of the present invention.

FIG. 2 is a diagram showing a memory map of various memories in FIG. 1;

FIG. 3 is a block diagram illustrating a configuration of a sound source unit in FIG. 1;

FIG. 4 shows one of four DSPs in FIG. 3;
FIG. 3 is a block diagram illustrating details of hardware of an SP.

FIG. 5 is a diagram for explaining a musical sound generation process performed by the sound source unit of FIG. 1;

FIG. 6 is a diagram for explaining in more detail a method of generating the basic musical sound waveform wavek and the musical sound waveform output choutk of FIG. 5 in one channel.

FIG. 7 is a flowchart showing a procedure of a main routine executed by the musical sound synthesizer of FIG. 1, in particular, a CPU;

FIG. 8 is a flowchart showing a detailed procedure of a tone color parameter setting processing subroutine of FIG. 7;

FIG. 9 is a flowchart showing a detailed procedure of a sound generation instruction processing subroutine of FIG. 7;

FIG. 10 is a diagram for explaining a process of giving an effect peculiar to an electromagnetic pickup of an electric piano.

FIG. 11 is a diagram illustrating an example of characteristics of a shift amount parameter that shifts a parameter Position according to a pitch.

FIG. 12 is a diagram for explaining a process of giving a specific effect in the relationship between the strings of the electric guitar and the electromagnetic pickup.

FIG. 13 is a diagram showing a string model and its equivalent conversion model.

FIG. 14 is a diagram for explaining a process of giving an effect of changing an input signal into a water sound.

FIG. 15 is a diagram for explaining a process of giving an effect similar to the PWM effect of the analog synthesizer.

FIG. 16 is a diagram illustrating a process of giving an effect generated by a flanger.

FIG. 17 is a diagram illustrating a process of giving an effect generated by a phaser.

FIG. 18 is a diagram for explaining a process of giving an effect of adding a harmonic component to an input signal.

FIG. 19 is a diagram for explaining a process of giving an effect by a non-carrier FM operator using an input signal as a modulator.

FIG. 20 is a diagram for explaining a process of giving an effect when an input signal is subjected to AM modulation by a signal of an internal oscillator.

FIG. 21 is a diagram for explaining a process of giving an effect when an input signal is output through an overdrive circuit after passing through a low boost filter.

[Explanation of symbols]

DESCRIPTION OF SYMBOLS 1 Keyboard (tone designation means) 2 Panel switch (tone designation means) 5 CPU (tone designation means) 16 Sound source section (basic waveform generation means, effect giving means, control means)

────────────────────────────────────────────────── ─── Continuation of the front page (56) References JP-A-8-54877 (JP, A) JP-A-6-138877 (JP, A) JP-A-62-194289 (JP, A) 174593 (JP, A) JP-A-5-46166 (JP, A) JP-A-8-110776 (JP, A) (58) Fields investigated (Int. Cl. ⁷ , DB name) G10H 1/02

Claims (6)

(57) [Claims] 1. A tone specifying means for designating a tone and a pitch of a tone to be generated, a basic waveform generating means for generating a basic tone waveform corresponding to the designated tone and pitch, and an effect algorithm program. by the execution, for the generation elementary tone waveform, the fundamental tone waveform
An effect imparting unit that imparts an effect including a process of delaying, and a control unit that controls the effect imparting unit so as to execute an effect algorithm program corresponding to the designated timbre. In order to give the effect peculiar to the electromagnetic pickup of the electric guitar,
An algorithm for changing a delay amount for delaying the basic musical sound waveform according to an assumed picking position of the electric guitar and an assumed position of the electromagnetic pickup, wherein the picking position and the pickup position are specified. A tone synthesizer characterized in that it is controlled in accordance with the pitch of a given tone. 2. A tone specifying means for designating a tone color and a pitch of a tone to be generated, a basic waveform generating means for generating a basic tone waveform according to the designated tone color and pitch, and an effect algorithm program. Executing the effect to the generated basic musical sound waveform, and control means for controlling the effect applying means to execute an effect algorithm program corresponding to the specified tone color. It has the effect algorithm generates a random signal, by controlling the cut-off frequency based on the random signal, bandpass Sufiru the該Ka <br/> cutoff frequency is randomly controlled
A tone synthesizer characterized in that the generated basic tone waveform is band-limited by a filter, and the band-limited tone waveform is injected into a loop in which at least the amount of delay is controlled according to the pitch. . 3. A tone specifying step for specifying a tone color and a pitch of a tone to be generated by the tone specifying means; a basic waveform generating step for generating a basic tone waveform corresponding to the specified tone and pitch; by executing the algorithm program, for the generation elementary tone waveform, the fundamental tone waveform
An effect imparting step of imparting an effect including a process of delaying, and a controlling step of controlling the effect imparting step so as to execute an effect algorithm program corresponding to the designated timbre, wherein the effect algorithm includes: In order to give the effect peculiar to the electromagnetic pickup of the electric guitar,
An algorithm for changing a delay amount for delaying the basic musical sound waveform according to an assumed picking position of the electric guitar and an assumed position of the electromagnetic pickup, wherein the picking position and the pickup position are specified. A tone synthesizing method characterized in that the tone is controlled in accordance with the pitch of the tone generated. 4. A tone specifying step for specifying a tone color and a pitch of a tone to be generated by the tone specifying means, a basic waveform generating step for generating a basic tone waveform in accordance with the specified tone and pitch, and An effect applying step of applying the effect to the generated basic tone waveform by executing an algorithm program; and a control of controlling the effect applying step to execute an effect algorithm program corresponding to the designated timbre. and a step, the effect algorithm generates a random signal, by controlling the cut-off frequency based on the random signal, bandpass fill 該Ka <br/> cutoff frequency is randomly controlled
A band which limits the generated basic musical tone waveform by means of a tone generator, and injects the band-limited musical tone waveform into a loop in which at least a delay amount is controlled according to the pitch. . 5. A tone designating module for designating a tone and pitch to be generated by tone designating means, a basic waveform generating module for generating a basic tone waveform in accordance with the designated tone and pitch, and an effect algorithm program. by the execution, for the generation elementary tone waveform, the fundamental tone waveform
An effect imparting module that imparts an effect including a process of delaying, and a control module that controls the effect imparting module to execute an effect algorithm program corresponding to the designated timbre, wherein the effect algorithm is an electric algorithm. To give the effect specific to the electromagnetic pickup of the guitar,
An algorithm for changing a delay amount for delaying the basic musical sound waveform according to an assumed picking position of the electric guitar and an assumed position of the electromagnetic pickup, wherein the picking position and the pickup position are specified. A storage medium storing a program that can be realized by a computer, wherein the program is controlled in accordance with a pitch of a generated musical tone. 6. A tone designating module for designating a tone and pitch to be generated by tone designating means, a basic waveform generating module for generating a basic tone waveform in accordance with the designated tone and pitch, and an effect algorithm program. Executing an effect algorithm module that applies the effect to the generated basic musical sound waveform, and a control module that controls the effect application module to execute an effect algorithm program corresponding to the specified timbre. wherein the said effect algorithm generates a random signal, by controlling the cut-off frequency based on the random signal, bandpass fill 該Ka <br/> cutoff frequency is randomly controlled
A band limited to the generated basic musical tone waveform by a data generator, and the band-limited musical tone waveform is injected into a loop in which at least a delay amount is controlled according to the pitch. A storage medium that stores programs that can be used.