JP3010693B2 - Musical sound wave generator - Google Patents

Description

Description: TECHNICAL FIELD The present invention relates to a sound source processing method in a musical tone waveform generator.

[Conventional technology]

Various electronic musical instruments with high performance have been realized by the development of digital signal processing technology and LSI processing technology. In particular, the emergence of an electronic musical instrument having a modulation type musical sound waveform generator that modulates an input signal generated based on performance information and obtains variously modulated waveforms and outputs the waveform as a musical sound waveform has been developed by professional amateurs. Regardless, it has contributed to a significant increase in the music population.

Since musical tone waveform generators for electronic musical instruments require a large amount of high-speed digital computation, they conventionally consist of a dedicated tone generator circuit that implements an equivalent hardware architecture of a tone generation algorithm based on the required tone generator system. ing. With such a sound source circuit, a sound source system based on a modulation system is realized.

The sound source circuits as described above have a large circuit scale in any of the sound source systems. In the case of an LSI, the size is about twice as large as that of a general-purpose data processing microprocessor even if a memory part for storing waveform data for modulation and the like is omitted. The reason is that in the tone generator circuit, complicated address control for accessing waveform data based on various performance information is required. Further, a register for temporarily holding intermediate data obtained in the process of the sound source generation processing needs to be disposed everywhere in an architecture corresponding to the sound source system. In particular, there are modulation methods in which the tone generation algorithm can be changed in various ways, and a hardware configuration corresponding to them is also required.

[Problems to be solved by the invention]

As described above, the conventional musical tone waveform generator is composed of a dedicated tone generator circuit compatible with the tone generator method, so the hardware scale becomes large, and the yield at the time of manufacturing an LSI chip in the case of realizing the LSI. In this respect, there is a problem that the cost is increased in the manufacturing stage and the musical tone waveform generator is increased in size.

In addition, when changing the tone generator system or changing the number of polyphonics, there is a problem that the tone generator circuit must be largely changed, resulting in an increase in cost in the development stage.

Further, when the conventional musical sound waveform generator is realized as an electronic device, data that can be processed by a tone generator circuit is generated from performance information corresponding to a performance operation, and communication of performance information with another musical instrument is performed. For example, a control circuit constituted by a microprocessor or the like is required. In such a control circuit, in addition to a performance information processing program for processing performance information, a sound source control program corresponding to a tone generator circuit for supplying data corresponding to the performance information to the tone generator circuit is required. In addition, it is necessary to operate both programs in synchronization. Due to the complexity of such a program, there is a problem that the cost is greatly increased in its development.

On the other hand, in recent years, many high-performance microprocessors for performing general-purpose data processing have been realized, and a musical sound waveform generator that performs sound source processing in software using such a microprocessor is realized. It is also possible. However, a technique for synchronizing a performance information processing program for processing performance information with a sound source processing program for executing sound source processing based on the performance information has not been known. In particular, since the processing time in the sound source processing program changes depending on the sound source system, a complicated timing control program for outputting the generated musical sound data to the D / A converter is required. As described above, if the sound source processing is simply performed in a software manner, the processing program becomes very complicated, and it is impossible to perform the processing of an advanced sound source system such as a modulation system in terms of processing speed and program capacity.

SUMMARY OF THE INVENTION It is an object of the present invention to enable advanced sound source processing by a modulation method under program control of a microprocessor without requiring a dedicated sound source circuit.

[Means for solving the problem]

The present invention has program storage means for storing a performance information processing program for processing performance information and a sound source processing program for obtaining a polyphonic tone signal in real time at a predetermined sampling period.

Next, there is provided an address control means for controlling an address of the program storage means.

Further, it has a data storage means for storing musical sound generation data necessary for generating the above-mentioned musical sound signal.

Further, there is provided an arithmetic processing means including a multiplier for executing the four arithmetic operations.

And it has a program execution means for executing the performance information processing program or the sound source processing program stored in the program storage means while controlling the above-mentioned address control means, data storage means and arithmetic processing means. The means normally causes the performance processing means to execute arithmetic processing according to each instruction of the performance processing program decoded by the decoding means, and generates musical sound generation data necessary for generating a musical sound signal. And storing the data in the data storage means, and transferring control to the sound source processing program at predetermined time intervals independent of the pitch of the musical tone to be generated,
According to each instruction of the sound source processing program decoded by the decoding means, the arithmetic processing means is caused to execute an arithmetic processing based on the musical sound generation data stored in the data storage means, thereby realizing a polyphonic music signal in real time. It is generated asynchronously with the predetermined time interval, and after the sound source processing program ends, the control is transferred to the performance processing program again. In this case, the program execution means has, for example, an interrupt control means for generating an interrupt signal at the predetermined time interval. Thereby, the program execution means interrupts the performance information processing program at the timing when the interrupt signal is generated from the interrupt control means during execution of the performance information processing program, transfers control to the sound source processing program and executes it. After completion, the interrupt is released and the execution of the performance information processing program is resumed.

In addition to the above configuration, a tone signal for temporarily holding the tone signal generated asynchronously with the predetermined time interval or the sampling period, and outputting the held tone signal in synchronization with the sampling period It has output means. The constant output time interval in this case is usually
This time interval is equal to the sampling period of the D / A converter or the like, but this time interval is the same as the above-mentioned predetermined time interval, or a tone signal for one sample is generated by executing the sound source processing program a plurality of times. In this case, the time interval is a multiple of the predetermined time interval.

[Action]

In the present invention, the program storage means, the address control means, the data storage means, the arithmetic processing means and the program execution means have the same configuration as a general-purpose microprocessor, and do not require any dedicated tone generator circuit. The tone signal output means has a configuration different from that of a general-purpose microprocessor, but is general-purpose in the category of a tone waveform generator.

As a result, the circuit scale of the entire musical tone waveform generator can be significantly reduced, and even in the case of LSI, the same manufacturing technology as that of a normal microprocessor can be used, and the yield of chips can be improved. Can be greatly reduced. Since the tone signal output means can be constituted by a simple latch circuit, there is almost no increase in manufacturing cost due to the addition of this part.

Further, when it is desired to change the modulation method between, for example, the phase modulation method and the frequency modulation method, or to change the number of polyphonics, it is possible to cope with it only by changing the sound source processing program stored in the program storage means. It is possible to greatly reduce the development cost of the device, and it is possible to provide a user with a new modulation method using, for example, a ROM card or the like.

The above operation is possible because the present invention has realized the following program architecture and data architecture.

That is, the present invention realizes a data architecture for storing tone generation data required for generating a tone by a modulation method on the data storage means. When the performance information processing program is executed, the tone generation data on the data storage means is controlled. When the sound source processing program is executed, a tone signal is generated based on the tone generation data on the data storage means. Is done. As described above, data communication between the performance information processing program and the sound source processing program is performed via the tone generation data on the data storage means, and the access to the data storage means in each program depends on the execution status of the partner program. , The two programs can be substantially configured as independent modules, and a simple and efficient program structure can be realized.

In addition to the above data architecture, in the present invention, normally, the performance information processing program is executed to perform, for example, scanning of keyboard keys and various setting switches, demonstration performance control, and the like. Is executed, and when the processing is completed, the program returns to the performance information processing program again. This allows the sound source processing program to forcibly interrupt the performance information processing program based on, for example, an interrupt signal generated at a predetermined time interval from the interrupt control means. There is no need to synchronize.

Further, when the program executing means executes the sound source processing program, the processing time varies depending on the type of modulation scheme or the selection of a tone generation algorithm in the modulation scheme, but this variation is entirely absorbed by the tone signal output means. be able to. Therefore, a complicated timing control program for outputting a tone signal to a D / A converter or the like is not required.

As described above, the data architecture in which the data link between the performance information processing program and the sound source processing program is performed through the tone generation data in the data storage means, and the sound source is generated at predetermined time intervals with respect to the performance information processing program. By realizing a program architecture for executing a processing program and further providing a tone signal output means, sound source processing based on efficient program control is realized with almost the same configuration as a general-purpose processor.

〔Example〕

 First, an outline of the present embodiment will be described.

FIG. 1 is an overall configuration diagram of the present embodiment, and shows an external memory.
The components other than 116 are implemented by one chip, and the two CPUs (central processing control units) of the master and slave among them exchange information with each other and perform sound source processing for musical tone creation.

For example, in the case of polyphonic of 16 channels, 8 channels are processed by the master CPU 101 and the remaining 8 channels are slaves.
Processing is performed by the CPU 102.

This sound source processing is performed by software, but in the case of performance, the sound source system of PCM or DPCM and FM
A sound source system according to a modulation system such as phase modulation or phase modulation is assigned.

This sound source method is automatically specified for a specific instrument tone, such as a trumpet or tuba,
For other instrument timbres, the sound source system can be selected by a selection switch, and can be automatically selected according to the musical range of the performance or the performance intensity such as key touch.

In addition, for example, it is also possible to assign different sound source schemes to two channels simultaneously with one key press, for example, the PCM scheme for the attack part and the FM scheme for the sustained part.

Furthermore, for example, in the case of the FM method, it takes too much time to perform processing by software according to the algorithm of the sound source processing, if it is performed by a general-purpose CPU, but this embodiment has solved that point.

 The above is the outline of the present embodiment.

Hereinafter, an embodiment of the present invention will be described with reference to the drawings.

In FIG. 1, first, an external memory 116 stores tone control parameters such as a target value of an envelope value, a tone waveform in a PCM (pulse code modulation) system, a tone difference waveform in a DPCM (differential pulse code modulation) system, and the like. It is remembered.

On the other hand, a master CPU (hereinafter abbreviated as MCPU) 101 and a slave CPU (hereinafter abbreviated as SCPU) 102 access the above data on the external memory 116 and perform sound source processing in a shared manner. These CPUs share the waveform data of the external memory 116, etc.
Contention may occur when reading data from 116. Therefore, each of the MCPU 101 and the SCPU 102 receives an external memory access address signal from the access address conflict avoidance circuit 105 via the MCPU external memory access address latch 103 and the SCPU external memory access address latch 104. By outputting the memory control data from the output terminals 111 and 112, the MCP
The conflict between the address from U101 and the address from SCPU 102 can be avoided.

Data read from the external memory 116 based on the address designation is input from the external memory data-in terminal 115 to the external memory selector 106. The external memory selector 106 converts the read data based on a control signal from the access address conflict avoidance circuit 105,
Data input to the MCPU 101 through the data bus MD,
The data is separated into data to be input to the SCPU 102 through the data bus SD and input to the MCPU 101 and the SCPU 102, respectively. Thus, data conflict can be avoided.

After that, the MCPU 101 and the SCPU 102 perform sound source processing on the respective data by software. After that, all the sound channels are accumulated, and the left output terminal 113 and the Right D terminal of the Left D / A converter 107 are accumulated. / A converter unit 108
From the light output terminal 114 of the
The left channel left analog output and the right channel right analog output are output.

FIG. 2 is a block diagram showing the internal configuration of the MCPU 101.

In the figure, a control ROM 201 stores a tone control program to be described later.
From 05, the program word (instruction) of the designated address is sequentially output via the ROM address decoder 202. Specifically, the word length of each program word is, for example, 28 bits, and a part of the program word is used as the lower part (in-page address) of the next address to be read out by the ROM address control unit 205.
The next address method is used for inputting the next address. Naturally, it may be constituted by a normal program counter type CPU.

The command analysis unit 207 analyzes an operation code of an instruction output from the control ROM 201, and sends a control signal to each unit of the circuit to execute a specified operation.

When the operand of the instruction from the control ROM 201 specifies a register, the RAM address control unit 204 specifies the address of the corresponding register in the RAM 206. The RAM 206 stores various tone control data, which will be described later with reference to FIGS. 11 and 12, for eight sounding channels, and also stores various buffers and the like which will be described later, and is used for sound source processing which will be described later.

When the instruction from the control ROM 31 is an arithmetic instruction, the ALU unit 208 and the multiplier 209 execute addition and subtraction and logical operation, and execute the multiplication based on an instruction from the command analysis unit 207.

The interrupt control unit 203, based on an internal hard timer (not shown), periodically updates the SCP of FIG.
A reset release signal A is supplied to U102, and an interrupt signal is supplied to the ROM address control unit 205 and each of the D / A converter units 107 and 108 in FIG.

In addition to the above-described configuration, the MCPU 101 in FIG. 2 is provided with the following various bus-related interfaces.

That is, in order to access the external memory 116, an interface 215 of an address bus MA for designating an address of the memory, and a data bus for exchanging the accessed data with the MCPU 101 via the external memory selector unit 106 The interface 216 of the MD, the interface 212 of the bus Ma for designating the address of the RAM inside the SCPU 102 to execute data transfer with the SCPU 102, and the MCPU 101
An interface 213 of a data bus Dout for writing data to the DCPU 102; an interface 214 of a data bus Din for the MCPU 101 to read data from the SCPU 102;
D / A data transfer bus interface 21 for transferring the final output waveform to A converter 107 and Right D / A converter 108
7, and an external switch or keyboard (see Fig. 8)
There are input / output ports 210 and 211 for exchanging data with the other.

 Next, the internal configuration of the SCPU 102 is shown in FIG.

Since the SCPU 102 only performs the sound source processing in response to the processing start signal from the MCPU 101, the SCPU 102 performs input / output for transmitting / receiving data to / from the interrupt control unit corresponding to FIG. 203 and the external circuits corresponding to FIGS. 210 and 211. Ports, and FIG.
D / A converter unit 107 and Right D / A converter unit 1 corresponding to
There is no interface for outputting a music signal in 08.
The other circuits 301, 302 and 304 to 309 are the same as those in FIG.
It has the same function as each of the circuits 1, 202, 204 to 209. Also,
Each of the interfaces 303, 310 to 313 is shown as 212 to 216 in FIG.
Are provided facing each other. The internal RAM address of the SCPU 102 specified from the MCPU 101 via the bus Ma is input to the RAM address control unit 304, and the corresponding address is specified for the RAM 306. Thus, for example, SCP
The accumulated waveform data for eight sounding channels generated by the U102 and held in the RAM 306 is stored in the M bus via the data bus DIN.
Output to CPU101. This will be described later.

In addition to the configuration shown above, in this embodiment, the MCPU 101
A function key 801 and a keyboard key 802 as shown in FIG. These parts constitute a substantial musical instrument operation unit.

Next, a D / A converter which is one of the features of the present invention will be described.

FIG. 6 (b) is a left / right D / A converter section of FIG.
This shows the internal configuration of 107 and 108 (the contents of both conversion units are the same). One sample data of a musical tone created by sound source processing is input to a latch 601 via a data bus. Then, when a sound source processing end signal is input to the clock input of the latch 601 from the command analysis unit 207 (FIG. 2) of the MCPU 101, musical sound data for one sample on the data bus is converted to the seventh sound data.
As shown in FIG.

Here, the time required for the sound source processing described above varies depending on the software for sound source processing. Therefore, when the sound source systems are different, the timing at which each sound source process ends and the tone data is latched in the latch 601 is not constant. Therefore, the output of the latch 601 cannot be directly input to the D / A converter 603 as shown in FIG.

Therefore, in this embodiment, as shown in FIG.
The output of (1) is further latched by the latch 602, and the tone signal is latched by the latch 602 by an interrupt signal equal to the sampling clock interval output from the interrupt control unit 203, and is output to the D / A converter 603 at regular intervals. I have to.

As described above, the use of the two latches absorbs a change in the processing time due to the sound source system, so that a complicated timing control program for outputting musical sound data to the D / A converter is unnecessary.

Next, the overall operation of the present embodiment will be described.

This embodiment is basically operate a center MCPU101, as shown in the main flow chart of the 4 (a), it is repeated a series of processes of S ₄₀₂ to S _410. The actual sound source processing is performed by interrupt processing. Specifically, the MCPU 101
And the SCPU 102 are interrupted, and based on the interrupt, each CPU performs a sound source process for generating eight channels of sound. When the processing is completed, the musical tone waveforms for a total of 16 channels, that is, 8 channels of each CPU, are added and output from the Left D / A converter unit 107 and the Right D / A converter unit 108. Thereafter, the process returns from the interrupt state to the main flow. Note that the above-described interrupt is periodically performed based on a hard timer in the interrupt control unit 203 in FIG. This period is equal to the sampling period at the time of outputting the musical sound.

The above is the schematic operation of this embodiment. Next, the operation of this embodiment will be described in detail with reference to FIG.

_S402 to _S402 in the main flowchart of FIG.
During the processing of S ₄₁₀ is repeatedly executed, when an interrupt from the interrupt control unit 203 is applied, MC of FIG (b)
The two processes of the PU interrupt process and the SCPU interrupt process of FIG. The “sound source processing” in FIGS. 7B and 7C is shown in FIG.

Now, the main flowchart of FIG. 4A shows a flow of processing other than the sound source processing, which is executed by the MCPU 101 in a state where no interrupt is issued from the interrupt control unit 203.

First, power is the ON, initial setting is performed, such as the contents of the RAM206 of MCPU101 (S _401).

Next, a function key connected to the outside of the MCPU 101, such as a tone switch shown in FIG. 27, is scanned ( _S402 ), and the state of each switch is read from the input port 210 to the RAM 20.
It is taken into the key buffer area in 6. As a result of the scanning, the function key whose state has changed is identified, and the processing of the corresponding function is performed ( _S403 ). For example, setting of a musical tone number, setting of an envelope number, and setting of a rhythm number if the additional function has rhythm performance.

Thereafter, the depressed keyboard key is captured in the same manner as in the case of the function key ( _S404 ), and a key assignment process is performed by identifying the changed key ( _S405 ).

Next, when the demonstration performance key is pressed with the function key 801 (see FIG. 8), the demonstration performance data (sequencer data) is sequentially read from the external memory unit 116, and key assignment processing and the like are performed ( S ₄₀₆ ). When the rhythm start key is pressed, the rhythm data is
6 are sequentially read, and key assignment processing and the like are performed ( _S407 ).

Thereafter, a timer process described below is performed (S
₄₀₈ ). That is, an interrupt timer process (S
The time value of the time data that has been incremented in step ₄₁₂ ) is determined, and is compared with sequencer data for time control sequentially read for demonstration performance control or rhythm data for time control read for rhythm performance control. , _S406 , or the rhythm performance of _S407 is performed.

Further, in the sound generation process _S409 , a pitch envelope process or the like is performed in which an envelope is added to the pitch of a musical tone to be sound-processed and pitch data is set in a corresponding sound generation channel.

Further, a flow one-round preparation process is executed ( _S410 ). In this process, the state of the sound channel of the note number at which the key is depressed in the keyboard key processing of _S405 is changed to the state of depressing the key, or the state of the sound channel of the note number of the released key is changed to the mute state. Are performed.

Next, the MCPU interrupt processing of FIG. 4B will be described.

When the interrupt control unit 203 of the MCPU 101 interrupts the MCPU 101, the processing of the main flowchart of FIG. 4A is interrupted, and the execution of the MCPU interrupt processing of FIG. 4B is started. In this case, in the MCPU interrupt processing program, control is performed so that the contents of registers and the like to which writing is performed in the main flow program of FIG. 4A are not performed. This eliminates the need for register saving and restoring processes performed at the start and end of the normal interrupt process. Thereby, the transition between the processing of the main flowchart of FIG. 4A and the interrupt processing of the MCPU is quickly performed.

Then, tone generator processing is started in MCPU interrupt processing (S _411). This sound source processing is shown in FIG.

Simultaneously with the above operation, the interrupt control unit of the MCPU 101
From 203 to the SCPU 102 ROM address control unit 305, the SCPU
A reset release signal A (see FIG. 1) is output, and SCPU1
At 02, the execution of the SCPU interrupt processing of FIG. 4 (c) is started.

Then, the sound source processing in the MCPU interrupt processing (S
₄₁₁₎ At about the same time, the sound source processing is started in the SCPU interrupt processing (S _415). As described above, both the MCPU 101 and the SCPU 102 simultaneously perform the sound source processing for each of the eight sounding channels, so that the processing time of the eight sounding channels can be reduced.
The sound source processing of 16 sounding channels can be executed, and the processing speed is approximately doubled (the above will be described later with reference to FIG. 5).

Subsequently, in MCPU101, after interrupt timer process S _412, waiting to come end signal SCPU interrupt processing from SCPU102 (S _413). In the interrupt timer process, the value of the time data (not shown) on the RAM 206 (FIG. 2) is incremented by utilizing the fact that the interrupt process of FIG. Is done. That is, the value of the time data indicates the passage of time. The time data thus obtained is used for time control in the timer processing _S408 of the main flow in FIG. 4A, as described above.

When tone generator processing of step S ₄₁₅ in the SCPU interrupt processing of FIG. 4 (c) is completed, the command analyzer 307 of SCPU102 the ROM address control unit 205 of the MCPU101, SC
The PU processing end signal B (see FIG. 1) is input. Thus, the determination of step S ₄₁₃ in the MCPU interrupt processing of FIG. 4 (b) is to YES.

As a result, the SCPU 102 via the data bus Din in FIG.
Waveform data generated in is loaded into RAM206 of MCPU101 (S _414). In this case, the waveform data is stored in the RAM 306 of the SCPU 102.
Since the command is stored in the upper predetermined buffer area (buffer B described later), the command analysis unit 207 of the MCPU 101
RAM address control unit 3 via internal addressing bus Ma
By specifying the above buffer address for 04, the waveform data is read.

Then, in S ₄₁₅ ′, the contents of the buffer area are
It is latched by the latch 601 (see FIG. 6) of the Left D / A converter unit 107 and Right D / A converter unit 108.

Next, using the flowchart of FIG.
The operation of the tone generator processing executed in step S ₄₁₅ of step S ₄₁₁ or SCPU interrupt processing of the CPU interrupt process will be described.

First, the waveform data addition area of the RAM 206 or the RAM 306 is cleared ( _S416 ). Then, tone generator processing is performed for each channel of the sound channel (S ₄₁₇ ~S _424), the end 8 channels th sound source processing 8 channels in a predetermined buffer area B at the time of the termination is added waveform Data is obtained. Details of these processes will be described later.

Next, FIG. 5 shows the aforementioned FIGS. 4 (a), (b),
5 is a flowchart conceptually showing the relationship between the processes in the flowchart of FIG. 5C, and shows how the MCPU 101 and the SCPU 102 perform sound source processing in a shared manner.

First, a certain process A (hereinafter, the same applies to B, C,..., F) is performed ( _S501 ). This “processing” corresponds to, for example, “function key processing” or “keyboard key processing” in the main flowchart of FIG. 4 (a). Then M
Enter CPU interrupt processing and SCPU interrupt processing,
The sound source processing is started by simultaneously MCPU101 and _{SCPU102 (S 502, S 503)} . When the SCPU interrupt processing in the SCPU 102 ends, the SCPU processing end signal B
Is input to In the MCPU interrupt processing, the sound source processing ends earlier than the SCPU interrupt processing, and waits for the end of the SCPU interrupt processing. When the SCPU processing end signal B is identified in the MCPU interrupt processing, the waveform data generated by the SCPU 102 is sent to the MCPU 101 and combined with the waveform data generated by the MCPU 101, and
It is output to ft D / A converter section 107 and Right D / A converter section 108. Then, some processing B in the main flowchart
Return to

The above-described operation is repeated while sound source processing is performed for all sounding channels (in this embodiment, 16 channels for the MCPU 101 and the SCPU 102) ( _S504 to _S504 ).
S ₅₁₆ ). This repetition processing is continued while the musical sound is being generated.

Data Configuration in Sound Source Processing Next, specific examples of the sound source processing executed in _S411 of FIG. 4B and _S415 of FIG. 4C will be described.

In the present embodiment, as described above, the CPUs of the MCPU 101 and the SCPU 102 share the sound source processing for each of the eight channels. As shown in FIG. 9, the data for the sound source processing for the eight channels is set in each area of the sound channels in the RAMs 206 and 306 of the MCPU 101 and the SCPU 102.

In addition, as shown in FIG. 12, BF, BT,
Buffers B and M are secured.

In this case, each sound source system can be set in each sound channel region of FIG. 9 by an operation described later in detail, as conceptually shown in FIG. 10, and when the sound source system is set, Data is set in each area of each sounding channel in FIG. 9 in a data format of each sound source system as shown in FIG. In this embodiment, as will be described later, it is possible to assign a different sound source system to each sounding channel.

In Table 1 showing the data format of each sound source system in FIG. 11, G is a sound source system number, which is a number for identifying the sound source system. The next A represents an address designated when waveform data is read during sound source processing,
A ₁ , A ₁ and A ₂ are the integer part of the current address and the external memory 11
It directly corresponds to the address where the waveform data of FIG. 6 (FIG. 1) is stored. Further, _AF is a decimal part of the current address, which is used for interpolation of the waveform data read from the external memory 116.

The following A _E is the end address, A _L respectively represent the loop address. Further, the integer part of the next P _I, P ₁ and P ₂ is the pitch data, P _F represents the fractional part of the pitch data. For example, P _I = 1 and P _F = 0 indicate the pitch of the original sound, P _I = 2 and P _F = 0.
Is the pitch one octave higher, and P _I = 0, P _F = 0.5
Represents a pitch one octave below.

The following is the X _P of the previous sample data, X _N represents the storage of the next sample data. Further, D represents a difference value of magnitude between two adjacent sample data, and E is an envelope value. Further, O is an output value, and C
Represents a flag used when changing a sound source system to be assigned to a sounding channel according to performance information, as described later.

Other various control data will be described later in the description of each sound source system.

Thus, the data as shown in FIG.
When a sound source system described later is determined and secured in the RAMs 206 and 306 of the SCPU 102, a channel
Data is set in the format shown in Figure 11.

Hereinafter, the sound source processing of each sound source system performed using such a data structure will be sequentially described. The sound source processing is performed by the command analysis unit of the MCPU 101 or the SCPU 102.
207 or 307 is realized by interpreting and executing a sound source processing program stored in the control ROM 201 or 301. Hereinafter, it is assumed that processing is performed on this premise unless otherwise specified.

First, in the flowchart of FIG. 4 (d), when entering each tone generator processing of each channel (one of S ₄₁₇ to S _424), in FIG. 11 which is stored in the corresponding sound channels of RAM206,306 The sound source method No. of the data of the data format (Table 1) shown is determined, and thereby, the sound source processing of the sound source method described below to be executed is determined.

When the sound source method No. indicates the PCM method, the sound source processing by the PCM method shown in the following operation flowchart of FIG. 13 is executed. Each variable in the flow is MCPU1
11 or the PC of the table 1 of FIG. 11 stored in the sound channel area of FIG. 9 on the RAM 206 or 306 of the SCPU 102.
Each data in M format.

FIG. 15 (a) shows an address at which the waveform data to be processed at present is stored among the addresses at which the PCM waveform data is stored on the external memory 116 (FIG. 1) (A _I , A _F ).

First, pitch data (P _I , P _F ) is added to the current address (S ₁₃₀₁ ). This pitch data corresponds to the type of the key that has been pressed with the keyboard key 801 or the like in FIG.

Then, whether the integer part A _I of the added address changes is determined (S _1302). Determination If NO, using the difference value D which is the difference between each sample data X _N and X _P in the address of the FIG. 15 (a) (A _I +1) and A _I, D × A _F becomes operational By the processing, the interpolation data value O corresponding to the decimal part _AF of the address is calculated ( _S1307 ). Incidentally, (see S ₁₃₀₆ to be described later) difference value D, which are Motoma' by tone generator processing in the current previous interrupt timing.

And this value is added the sample data X _P corresponding to the integer part A _I address to the interpolation data value O is the current address (A _I, A _F) new sample data corresponding to the O of (FIG. 15 (a) X _Q ) is obtained ( _S1308 ).

Thereafter, the sample data is multiplied by an envelope value E (S ₁₃₀₉ ), and the contents of the obtained O are stored in the waveform data buffer B in the RAM 206 or 306 of the MCPU 101 or SCPU 102.
(See FIG. 12) (S ₁₃₁₀ ).

Thereafter, the process returns to the main flow of 4 (a), takes interrupt the sampling period of the next, is executed operation flowchart of tone generator processing of Fig. 13 is again, the current address (A _I, A _F) the pitch data (P _I , P _F ) are added (S ₁₃₀₁ ).

The above operation is repeated until the integer part A _I of address changes (S _1302).

During this time, the sample data _XP and the difference value D are not updated, only the interpolation data O is updated according to the address _AF ,
Each time, sample data _XQ is obtained.

Next, as a result of adding the pitch data (P _I , P _F ) to the current address (A _I , A _F ) in S ₁₃₀₁ , the integer part A of the current address is obtained.
_{If I} changes ( _S1302 ), address A _I is the end address
It is determined whether _AE has been reached or _exceeded ( _S1303 ).

If the determination is YES, the next loop processing is performed. That is, the address exceeding the end address A _E (A _I −
A _E ) is added to the loop address A _L , and loop reproduction is started from the integer part A _{I of the} obtained new current address (S
₁₃₀₄ ). The end address _AE is an external memory 116 in which the last waveform sample data of the PCM waveform data is stored.
(FIG. 1). Also, the loop address
A _L is an address of a position where the player wants to repeat the output of the waveform, and the above operation realizes a well-known loop process in the PCM system.

If the determination in _S1303 is NO, the processing in _S1304 is not executed.

Next, the sample data is updated. Here, from the external memory 116 (Figure 1) is read each sample data corresponding to the newly updated current address A _I and one before the address (A _I -1), respectively X _N, as X _P ( _S1305 ).

Furthermore, the difference value so far is updated to the difference value D between the X _N and X _P the updated (S _1306).

 The subsequent operation is as described above.

As described above, waveform data for one sounding channel is generated by the PCM method.

Sound source processing by DPCM system Next, sound source processing by the DPCM system will be described.

First, the operation principle of the DPCM method will be outlined with reference to FIG. 15 (b).

In the figure, address A of external memory 116 (FIG. 1)
Sample data X _P corresponding to _I, the previous one of the addresses A _I, especially is a value determined from the differential value between the sample data corresponding to the not-shown address (A _I -1).

Since a difference value D from the next sample data is written in the address A _I of the external memory 116 (FIG. 1),
Sample data of the next address Motomari in X _P + D, which replaces the new sample data X _P.

In this case, assuming that the current address is A _{F as} shown in the figure, the sample data corresponding to the current address A _F is X _P
+ D × A _F

As described above, in the DPCM method, the difference value D between the current address and the sample data corresponding to the next address is read from the external memory 116 (FIG. 1), added to the current sample data, and Is obtained, waveform data is sequentially created.

When such a DPCM method is used, when quantizing a waveform such as a voice or a musical tone in which the difference value between adjacent samples is generally small, the quantization is performed with a much smaller number of bits as compared with the normal PCM method. Obviously,

The operation of the above DPCM method will be described with reference to the operation flowchart of FIG. Each variable in the flow is each data in the DPCM format of Table 1 in FIG. 11 stored in one of the sounding channel areas in FIG. 9 on the RAM 206 or 306 of the MCPU 101 or the SCPU 102.

FIG. 15 (b) shows an address at which data to be processed at present is stored among addresses at which DPCM differential waveform data is stored on the external memory 116 (FIG. 1) (A _I , A _F ).

First, pitch data (P _I , A _F ) is added to the current address (A _I , A _F ).
P _F) is added (S _1401).

Then, it is determined whether there is a change in the integer part A _I of the added address ( _S1402 ). If the judgment is NO, the 15th
Using the difference value D at the address A _I in FIG.
The A _F becomes the arithmetic processing, the interpolation data values O corresponding to the decimal part A _F address is calculated (S _1414). Incidentally, the difference value D (see S ₁₄₀₆ and S ₁₄₁₀ to be described later) which are Motoma' by tone generator processing in the current previous interrupt timing.

Then are added the sample data X _P corresponding to the integer part A _I address to the interpolation data value O is, current address (A _I, A _F) new sample data O corresponding to (FIG. 15 (b), X _Q ) is obtained (S ₁₄₁₅ ).

Thereafter, the sample data is multiplied by an envelope value E (S ₁₄₁₆ ), and the contents of the obtained O are stored in the waveform data buffer B in the RAM 206 or 306 of the MCPU 101 or SCPU 102.
(See FIG. 12) (S ₁₄₁₇ ).

Thereafter, the process returns to the main flow of 4 (a), takes interrupt the sampling period of the next, is executed operation flowchart of tone generator processing of FIG. 14 is again, the current address (A _I, A _F) the pitch data (P _I , P _F ) are added (S ₁₄₀₁ ).

The above operation is repeated until a change occurs in the integer part A _I of the address.

During this time, the sample data _XP and the difference value D are not updated, only the interpolation data O is updated according to the address _AF ,
Each time, new sample data _XQ is obtained.

Next, the pitch data (P _I , P _F ) is added to the current address (A _I , A _F ) in S ₁₄₀₁ , and as a result, the integer part A of the current address is obtained.
_{If I} changes ( _S1402 ), address A _I is the end address
It is determined whether _AE has been reached or _exceeded ( _S1403 ).

If the determination is NO, the loop processing of the following S ₁₄₀₄ to S _1407, the sample data currently corresponding to the integer part A _I of the address is calculated. That is, first, a variable called old A _I (the
11 the reference column of the DPCM table 1) of the figure, the value before the current integer part A _I address changes are stored. This is achieved by repeating the processing of S ₁₄₀₆ or S ₁₄₁₃ will be described later. While the value of the old A _I is sequentially incremented at S _1406, the difference waveform data in the external memory 116 (FIG. 1) indicated by old A _I in S ₁₄₀₇ is read as D, sequentially samples in S ₁₄₀₅ It is accumulated in the data X _P. The integer part of the current address after the value of the old A _I change
When it becomes equal to A _I, the value of the sample data X _P is the current value corresponding to the integer part A _I address after the change.

In this way, the sample data X _P currently corresponding to the integer part A _I address is obtained, the determination is YES in S _1404, proceeds to the processing of the aforementioned interpolated value (S _1414).

Tone generator processing described above is repeated for each interrupt timing, the determination of S ₁₄₀₃ is When changes to YES, enters a loop processing of the next.

First, addresses exceeding the end address A _E (A _I −
A _E ) is added to the loop address A _L , and the obtained address is used as the integer part A _I of the new current address (S ₁₄₀₈ ).

Hereinafter, the how advanced address from the loop address A _L, by operation of accumulating several times the difference value D is repeated, the sample data X _P corresponding to the integer part A _I of the new current address is calculated Is done. That is, first, the sample data X _PL (first in the loop address A _L that sample data X _P as the initial setting is set in advance
(See the DPCM column in Table 1 in Figure 1)
Old A _I is the value of the loop address A _L (S _1409). Then, the process is repeated in S ₁₄₁₀ to S ₁₄₁₃ described below. That is, while the values of the old A _I is sequentially incremented at S _1413, an external memory 116 (first indicated by old A _I in S ₁₄₁₀
Differential waveform data on the Figure) is read as D, S ₁₄₁₂
It is accumulated sequentially sample data X _P at. And
When the value of the old A _I is equal to the integer portion A _I of the new current address, the value of the sample data X _P becomes a value corresponding to the integer part A _I of the new current address after loop processing.

In this way, the sample data X _P corresponding to the integer part A _I of the new current address is obtained, the determination of S ₁₄₁₁ is YES
Then, the processing shifts to the above-described interpolation value calculation processing ( _S1414 ).

As described above, waveform data for one sounding channel is generated by the DPCM method.

Sound source processing by FM modulation method (No. 1) Next, sound source processing by FM modulation method will be described.

In the FM modulation method, usually, the same hardware or software called an operator as shown by OP1 to OP4 in FIG. 18 is used, and they are connected as shown in algorithms 1 to 4 in FIG. Musical sounds are generated by being connected to each other by rules. In this embodiment, the FM modulation method is realized by software.

Next, the operation of one embodiment in the case where sound source processing is performed by two operators will be described with reference to the operation flowchart of FIG. 16 (a). The processing algorithm is shown in FIG. Each variable in the flow is MCPU101 or SCP
These are the data in the FM format of Table 1 in FIG. 11 stored in any of the sounding channel areas in FIG. 9 on the RAMs 206 and 306 of U102.

First, the processing of the operator 2 (OP2), which is a modulator, is performed. For pitch processing, since the interpolation is not performed as PCM method, only integer address A _2. That is, it is assumed that the waveform data for modulation is stored in the external memory 116 (FIG. 1) at sufficiently small step intervals.

First, the pitch data P ₂ is added current to the address A ₂ (S _1601).

Then, the address A ₂ feedback output F _O2 is added as a modulation input, a new address A _M2 is obtained (S _1602). Feedback output F _O2 are obtained by treatment of S ₁₆₀₅ to be described later in the previous interrupt timing is executed.

Further, the value of the sine wave corresponding to the address A _M2 (phase) is calculated. Actually, the sine wave data is stored in the external memory 116 (FIG. 1), and the sine wave data is obtained by looking up the sine wave data at the address _AM2 ( _S1603 ).

Subsequently, output O ₂ envelope value E ₂ is multiplied by the sine wave data is obtained (S _1604).

Thereafter, the feedback level F _L2 is multiplied feedback output F _O2 is obtained on the output O ₂ (S _1605).
In the case of the present embodiment, this output _FO2 is input to the operator 2 (OP2) at the next interrupt timing.

Also, modulation level M _L2 to O ₂ is multiplied modulation output M _O2 is obtained (S _1606). The modulation output M _O2 will modulation input to the operator 1 (OP1).

Next, the processing shifts to the processing of the operator 1 (OP1). This process is similar to the following, except that there is no modulation input by the feedback output.
This is almost the same as the case of the operator 2 described above.

First, the pitch data in the current address A ₁ of the operator 1
P ₁ is added (S _1607), a new address A _M1 is obtained modulation output M _O2 above this value is added (S _1608).

Next, the value of the sine wave corresponding to the address A _M1 (phase) is read from the external memory 116 (FIG. 1) (S ₁₆₀₉ ), and is multiplied by the envelope value E ₁ to generate the musical sound waveform output O _1. Obtained ( _S1610 ).

And this is the RAM 206 (FIG. 2) or 306 (FIG. 3)
_Is added to the buffer B (see FIG. 12) in (S ₁₆₁₁ ),
The FM modulation processing for one sounding channel ends.

Sound source processing by TM modulation method (No. 1) Next, sound source processing by the TM modulation method will be described.

 First, the principle of the TM modulation method will be described.

FM modulation method described above is based on the _{e = A · sin {ω c} t + I (t) · sinω m t} becomes expression. However, where, omega _c t is the carrier phase angle (carrier signal), the sin .omega _m t modulation wave phase angle (modulation signal), and I (t) is the modulation index.

In contrast, phase modulation system referred to as TM modulation method in this embodiment, a basic _{e = A · f T {f} c (t) + I (t) · sinω m t} becomes expression. Here, f _T (t) is a triangular wave function, and is defined by the following function for each region of each phase angle (however, ω is input).

f _T (ω) = 2 / π · ω (region: 0 ≦ ω ≦ π / 2) f _T (ω) = − 1 + 2 / π (3π / 2−ω) ·· (region: π / 2 ≦ ω ≦ 3π / 2) f _T (ω) = − 1 + 2 / π (ω-3π / 2) (region: 3π / 2 ≦ ω ≦ 2π) Further, f _c is called a modified sine wave, and each phase is This is a carrier signal generation function obtained by accessing the external memory 116 (FIG. 1) in which different sine waveform data is stored for each corner region at the carrier phase angle ω _ct . F _{c for} each region of each phase angle
Is defined as

_{f c (t) = π /} 2sinω c t ·· ( area: 0 ≦ ωt ≦ π / 2 ) f c (t) = π-π / 2sinω c t ·· ( area: π ≦ ωt ≦ 3π / 2 ) _{f c (t) = 2π +} π / 2sinω c t ··: in (region _{3π / 2 ≦ ω c t ≦} 2π) ( where n is an integer) TM modulation scheme is generated in such the above function f _c (t) The above-described triangular wave function is modulated by an addition signal obtained by adding the modulation signal sinω _mt to the carrier signal at a rate indicated by the modulation index I (t). Thus, the modulation index I (t)
If the blood is zero, a sine wave can be generated and I
By increasing the value of (t), a very deeply modulated waveform can be generated. Where the modulation signal sinω _m t
Instead, various signals can be used. As described below, it is possible to feed back the own operator output at the previous calculation at a fixed feedback level or to input the output of another operator.

Sound source processing based on the TM modulation method based on this principle is described in the 17th
This will be described with reference to the operation flowchart of FIG. This case is also an example in which the sound source processing is performed by two operators, as in the case of the FM modulation method in FIG. 16, and the processing algorithm is shown in FIG. 17 (b). Each variable in the flow is each data in the TM format of Table 1 in FIG. 11 stored in any of the sounding channel areas in FIG. 9 on the RAM 206 or 306 of the MCPU 101 or the SCPU 102.

First, the processing of the operator 2 (OP2), which is a modulator, is performed. For pitch processing, since the interpolation is not performed as PCM method, only integer address A _2.

First, the pitch data P ₂ is added current to the address A ₂ (S _1701).

Then, the deformation sine transform f _c, the address A
_The modified sine wave corresponding to ₂ (phase) is read from the external memory 116 (FIG. 1), and the carrier signal is generated as O ₂ (S ₁₇₀₂ ).

Subsequently, the feedback output F _O2 (S ₁₇₀₆ ) is added as a modulation signal to the above-described O ₂ which is a carrier signal, and a new address is obtained and is set as O ₂ (S ₁₇₀₃ ). Feedback output F _O2 are obtained by treatment of S ₁₇₀₆ to be described later in the previous interrupt timing is executed.

Then, the value of the triangular wave corresponding to the added address O ₂ described above is calculated. Actually, the above-mentioned triangular wave data is stored in the external memory 116 (FIG. 1),
In O ₂ obtained by the triangular wave data table pull (S _1704).

Subsequently, output O ₂ envelope value E ₂ is multiplied by the triangular wave data is obtained (S _1705).

Thereafter, the feedback level F _L2 is multiplied feedback output F _O2 is obtained on the output O ₂ (S _1707).
In the case of the present embodiment, this output _FO2 is input to the operator 2 (OP2) at the next interrupt timing.

Also, modulation level M _L2 to O ₂ is multiplied modulation output M _O2 is obtained (S _1707). The modulation output M _O2 will modulation input to the operator 1 (OP1).

Next, the processing shifts to the processing of the operator 1 (OP1). This process is similar to the following, except that there is no modulation input by the feedback output.
This is almost the same as the case of the operator 2 described above.

First, the pitch data in the current address A ₁ of the operator 1
P ₁ is added (S ₁₇₀₈ ), and the obtained value is subjected to the above-described modified sine transform to obtain a carrier signal as O ₁ (S ₁₇₀₉ ).

Next, the O ₁ of the above modulation output M _O2 are the new O ₁ is added (S _1710), the value O ₁ is a triangular wave conversion (S _1711), and is further multiplied envelope values E ₁ is tone waveform output O ₁ is obtained (S _1712).

This is added to the buffer B (see FIG. 12) in the RAM 206 (FIG. 2) or 306 (FIG. 3) (S ₁₇₁₃ ), and the TM modulation processing for one tone generation channel is completed.

The sound source processing according to the four methods of PCM, DPCM, FM, and TM has been described above. Among these, the two systems of FM and TM are modulation systems, and in the above-described examples, both are processes by two operators based on the algorithm shown in FIGS. 16 (b) and 17 (b). The sound source processing at the time of performance has more operators and the algorithm is more complicated. An example is shown in FIG. In the algorithm 1 shown in the figure, quadruple modulation including feedback input is performed.
A complicated waveform can be obtained. In Algorithm 2, two sets of algorithms having a feedback input are arranged in parallel, and are suitable for expressing, for example, a timbre change when shifting from attack to sustain. Algorithm 4
Have characteristics close to the sine wave synthesis method.

An example of normally considered sound source processing of the FM system and the TM system by the four operators in FIG. 18 will be sequentially described with reference to FIGS. 19 and 20.

Sound source processing by FM modulation method (No. 2) FIG. 19 is an operation flowchart of sound processing that can be normally considered based on the FM modulation method corresponding to the algorithm 1 in FIG. Each variable in the flow is MCPU101 or SCPU1
02 is stored in any of the sounding channel areas shown in FIG. Although it does not correspond to each data of the FM format of the table 1 in FIG. 11, it is obvious that it can be easily realized by extending the data format.

First, to the current address A ₄ of the operator 4 (OP4),
Pitch data P ₄ is added (S _1901). Next, the address A ₄ to the feedback output F _{O4 (1905)} is added as a modulation input, a new address A _M4 is obtained (S
₁₉₀₂ ). Furthermore, the value of the sine wave corresponding to the address A _M4 (phase) is read from the external memory 116 (see FIG. 1) (S _1903), this is multiplied envelope value E ₄ is the output O ₄ is obtained ( _S1904 ). Then, it is multiplied to the output O ₄ feedback level F _L4 feedback output F
_O4 is obtained (S _1905). Also, modulation level M _L4 is multiplied to the output O ₄ with modulation output M _O4
Is obtained ( _S1906 ). This modulation output M
_O4 is a modulation input to the next operator 3 (OP3).

Next, the processing shifts to the processing of the operator 3 (OP3). This process is similar to the following, except that there is no modulation input by the feedback output.
This is almost the same as the case of the operator 4 described above. First, the pitch data P _{3 is} assigned to the current address A ₃ of the operator 3 (OP3).
Is added ( _S1907 ). Then, the modulation output M _O4 is added as a modulation input to the address A _3, a new address A _M3 is obtained (S _1908). Further this address
A The value of the sine wave corresponding to _M3 (phase) is
Read from (FIG. 1) (S _1909), which is the output O ₃ envelope value E ₃ are multiplied to obtain the (S _1910). Thereafter, the modulation level M _L3 is multiplied modulation output _O3 is obtained on the output O ₃ (S _1911). This modulation output _MO3 becomes a modulation input to the next operator 2 (OP2).

Next, the process of the operator 2 (OP2) is executed. This process is the same as that of the above-described operator 3 except that the modulation input is different, and thus the description is omitted.

Finally, although starts the processing of the operator 1 (OP1), which is also to step S _1920, to date the same processing is performed. Then, tone waveform output O ₁ obtained in S ₁₉₂₀ is accumulated in the buffer B (see FIG. 12) as the carrier (S _1921).

Sound source processing by TM modulation method (No. 2) FIG. 20 is an operation flowchart of sound processing that is normally considered based on the TM modulation method corresponding to the algorithm 1 of FIG. Each variable in the flow is MCPU101 or SCPU1
02 is stored in any of the sounding channel areas shown in FIG. It should be noted that it does not correspond to each data of the TM format of the table 1 of FIG. 11, but it is obvious that it can be easily realized by extending the data format of FIG.

First, the pitch data P ₄ is added to the current address A ₄ of the operator _{4 (OP4) (S 2001)} . Then, the deformation sine transform f _c, deformation sine wave corresponding to the above address A ₄ (phase) is read from the external memory 116 (Figure 1), the carrier signal is generated as O ₄ (S ₂₀₀₂ ). The output O ₄ to the feedback output as a modulation signal F _O4 (S
₂₀₀₇ reference) is added, the new address O ₄ is obtained (S _2003). Then, the value of the triangular wave corresponding to the address O ₄ (phase) is read from the external memory 116 (Figure 1) (S _2004), this is multiplied envelope value E ₄ is the output O ₄ is obtained (S ₂₀₀₅ ). Thereafter, the output O ₄ is multiplied by the modulation level M _L4 to obtain a modulation output M _O4 (S ₂₀₀₆ ), and the output O ₄ is multiplied by the feedback level _{FL 4} to obtain the feedback output F _L4.
O4 is obtained (S _2007). The modulation output M _O4
Is the modulation input to the next operator 3 (OP3).

Next, the processing of the operator 3 (OP3) is started. This process is similar to the following, except that there is no modulation input by the feedback output.
This is almost the same as the case of the operator 4 described above. First, is added to the current pitch data P ₃ to the address A ₃ operators 3 (S _2008), it is performed modified sine transform the obtained value carrier signal is obtained as O ₃ (S _2009). Then, the O ₃ to the aforementioned modulation output M _O4 is added is a new O ₃ (S _2010), the value O ₃ is a triangular wave conversion (S _2011), and is further multiplied by the envelope values E ₃ the output O ₃ obtained (S _2012). Further, this modulation level M _L3 is multiplied modulation output M _O3 is obtained (S _2013). This modulation output
_MO3 is a modulation input to the next operator 2 (OP2).

Next, the process of the operator 2 (OP2) is executed. This process is the same as that of the above-described operator 3 except that the modulation input is different, and thus the description is omitted.

Finally, the process of the operator 1 (OP1) is started, and the same process as before is performed up to step _S2024 . Then, tone waveform output O ₁ obtained in S ₂₀₂₄ is accumulated in the buffer B (see FIG. 12) as the carrier (S _2025).

As described above, the embodiment of the normal sound source processing by the modulation method has been described. However, this processing is processing for one sounding channel as described above, and is actually performed for eight sounding channels for each CPU of the MCPU 101 and the SCPU 102. (See FIG. 4 (d)). If the modulation method is specified for a certain sound channel, the sound source processing by the above-described modulation method operates.

Modification Example (Part 1) Next, a first modification example of the sound source processing by the modulation method will be described.

The basic concept is shown in the flowchart of FIG.

In the figure, operators 1, 2, 3, and 4 have the same program structure except for the variable names used.

Here, each operator process cannot perform the operator process unless the modulation input is determined. This is because the modulation input to each operator process differs depending on the algorithm as shown in FIG. In other words, which operator process output is used as the modulation input,
Alternatively, it is necessary to feed back the output of one's own operator's processing instead of the other operator's processing and determine whether to use the output of one's own modulation. Therefore, in the operation flow of FIG. 21, such a connection relationship algorithm processing done collectively as (S _2105), the operator process connection relation thus obtained is in the next interrupt timing (S ₂₁₀₂ ~S ₂₁₀₄₎ To determine the modulation input at. It should be noted that an initial value is given as an input to each operator process at the start of sound generation (that is, at the beginning).

As described above, if the operator process and the algorithm process are separated, the program of the operator process may be the same for any algorithm, and only the algorithm process may be changed. Therefore, it is possible to greatly reduce the program capacity of the entire sound source processing by the modulation method.

Next, an improved example of the FM modulation method based on the above-described basic concept will be described. FIG. 21 (a) is a flowchart showing the operation of the operator process shown in FIG. 21 using the FM modulation method, and FIG. 22 (b) shows the calculation algorithm per operator, taking the operator 1 process as an example. Other operators 2
The four processes are exactly the same except that the subscript numbers of the variables are different. Each variable in the flow is the MCPU 101 or the SCPU 102
Is stored in any one of the tone generation channels shown in FIG.

First, the pitch data P _{1 is} stored in the address A ₁ corresponding to the phase angle.
There are added, it is a new address A ₁ (S _2201). The modulated input M _I1 to this address A ₁ is added, the address A _M1 is obtained (S _2202). Modulation input M _I1 is the algorithm processing S at the previous interrupt timing.
₂₁₀₅ (FIG. 21), depending on the algorithm, the operator's own feedback output F _O1 or another operator, eg operator 2
_Or the output _MO2 . Next, the value of the sine wave corresponding to the address (phase) A _M1 is stored in the external memory 116 (first memory).
Read from the drawing), the output O ₁ is obtained (S _2203). After that, the value obtained by multiplying this by the envelope data E ₁ is
It becomes the output O ₁ of the operator 1 (S _2204). Further, the output O ₁ to be multiplied feedback level F _L1 is the feedback output F _O1 obtained (S _2205), also modulation level M _L1 is multiplied to the output O _1, the modulation output M _O1 is obtained ( _S2206 ).

Next, an improved example of the TM modulation scheme based on the above-described basic concept will be described. FIG. 23 (a) shows an operation flowchart of the TM modulation method of the operator processing in FIG. 21 taking the operator 1 processing as an example, and FIG. 21 (b) shows the calculation algorithm per operator. Other operators 2
The four processes are exactly the same except that the subscript numbers of the variables are different. Each variable in the flow is the MCPU 101 or the SCPU 102
Is stored in any one of the tone generation channels shown in FIG.

First, the pitch data P ₁ is added to the current address A ₁ (S _2301). Then, the deformation sine transform f _c, deformation sine wave corresponding to the above address A ₁ (phase) is read from the external memory 116 (Figure 1), the carrier signal is generated as O ₁ (S ₂₃₀₂ ). The modulation input M _I1 as a modulation signal is added to the output O _1, a new address O ₁ is obtained (S _2303). Next, the value of the triangular wave corresponding to the address O ₁ (phase) is read from the external memory 116 (S
_2304), the envelope values E ₁ is the output O ₁ is multiplied thereto (S _2305). Thereafter, the feedback level F _L1 is multiplied feedback output F _O1 is obtained at the output O ₁ (S _2306), also modulation level M _L1 is multiplied to the output O ₁ modulation output M _O1 and is obtained ( _S2307 ).

Next, a specific example of the algorithm processing _{S2105 in} FIG. 21 for determining the modulation input in the operator processing in the above-described FM and TM modulation schemes will be described with reference to the operation flowchart in FIG. The flow of FIG. 13 is common to the FM method and the TM method, and is an example in which processing is performed by switching the algorithms 1 to 4 in FIG. In this case, the selections of the algorithms 1 to 4 are selected based on an instruction (not shown) by the player ( _S2400 ).

First, as shown in FIG.
The operator (abbreviated as OP) is of a serial type, and only OP4 has a feedback input. That is, the feedback output F _O4 of OP4 is set as the modulation input M _{I4 of} OP4 ( _S2401 ), the modulation output M _{O4 of} OP4 is set as the modulation input M _{I3 of} OP3 (S ₂₄₀₂ ), and the modulation output M _{O3 of} OP3 is set as OP2. is a modulation input M _I2 (S _2403), the modulation output M _O2 of OP2 is set to the modulation input M _I1 of OP1 (S _2404), the buffer B (12 output O ₁ of OP1 as the carrier output
(Refer to the figure) ( _S2405 ).

Algorithm 2 is composed of OP2 and OP4, as shown in FIG.
Have a feedback input. That is, the feedback output F _O4 of OP4 is set as the modulation input M _{I4 of} OP4 ( _S2406 ), the modulation output M _{O4 of} OP4 is set as the modulation input M _{I3 of} OP3 (S ₂₄₀₇ ), and the feedback output F _O2 of OP2 is set as OP2. is a modulation input M _I2 (S _2408), OP2 and modulation output M _O2, M _O3 of OP3 is the modulation input M _I1 of OP1 (S _2409), the output O ₁ of the OP1 is added to the buffer B as the carrier output (S ₂₄₁₀ ).

In Algorithm 3, OP2 and OP4 have feedback inputs, and two 2-operator serial types are configured in parallel. In other words, the feedback output F _O4 of OP4 is the modulation input M _I4 of OP4 (S _2411), the modulation output M _O4 of OP4 is the modulation input M _I3 of OP3 (S _2412), the feedback output F _O2 of OP2 is OP2 The modulation output M _{O2 of} OP2 is used as the modulation input M _I2 (S ₂₄₁₃ ), the modulation output M _O2 of OP ₂ is used as the modulation input M _I1 (S ₂₄₁₄ ), and the outputs O ₁ and O _{3 of} OP 1 and OP ₃ are used as carrier outputs in buffer B. _Is added ( _S2415 ).

Algorithm 4 is a 4-operator parallel type, where all operators have feedback inputs. That is, the feedback output _FO4 of OP4 is set as the modulation input M _{I4 of} OP4 ( _S2416 ), the feedback output _FO3 of OP3 is set as the modulation input M _{I3 of} OP3 ( _S2417 ), and the feedback output _FO2 of OP2 is set as OP2. The modulation input M _I2 of (S ₂₄₁₈ ), the feedback output _{FO 1} of OP ₁ is used as the modulation input M _{I1 of} OP 1 (S ₂₄₁₉ ), and the outputs O ₁ , O ₂ , O ₃ and O _{4 of} all operators are buffer B (S ₂₄₂₀ ).

The sound processing for one channel is completed by the operator processing and the algorithm processing described above, and if there is no change in the algorithm, the sound is generated as it is (sound processing).
Followed by

Next, a second improved example of the sound source processing by the modulation method will be described.

In the various modulation methods described so far, the more complicated the algorithm is, and the more the number of sounding channels (the number of polyphonics), the longer the processing time.

Therefore, in the second improved example described below, the first improved example in FIG. 21 in which the operator processing and the algorithm processing are separated is further developed, and only the operator processing is performed at a certain interrupt timing, and the next interrupt timing is performed. Then, the operator process and the algorithm process are alternately performed such that only the algorithm process is performed.
As a result, the processing load per one interrupt timing can be significantly reduced. As a result, one sample data is output every two interrupts.

This operation will be described with reference to the operation flowchart of FIG.

First, in order to alternately perform the operator process and the algorithm process, it is determined whether or not the value is zero using the variable S (S
₂₅₀₁ ). This variable S is provided for each sounding channel,
It is stored in each sounding channel area of FIG. 9 on the RAM 206, 306 of the MCPU 101 or SCPU 102.

At a certain interrupt timing, when S is 0, the process enters the route of the operator processing, and the variable S is set to a value of 1 ( _S2502 ). Subsequently, the operator 1-4 processing is executed (S ₂₅₀₃ ~S _2506). This processing is performed in accordance with FIG.
It is the same as FIG.

Next, the operator exits the route of the operator processing, and
Then, an output process in which the value of the buffer BF (for the FM system) or the value of the buffer BT (for the TM system) is set (S
₂₅₁₀ ). The buffer BF or BT is provided for each sounding channel, and is stored for each sounding channel area in FIG. 9 on the RAM 206, 306 of the MCPU 101 or SCPU 102. Buffer BF or
The waveform output value after the algorithm processing is stored in BT, but at the current interrupt timing, the algorithm processing has not been executed and the contents of the buffer BF or BT have not been changed, so the same as the previous interrupt timing The waveform output value is output.

With the above processing, the sound source processing for one sounding channel at the current interrupt timing is completed. In this case, each data obtained by the current processing of the operators 1 to 4 is transferred to the MCPU 101 or SCPU until the next interrupt timing.
It is stored in each sounding channel area of FIG. 9 on the RAMs 206 and 306 of the CPU 102.

When the next interrupt occurs, since the variable S has been set to 1 at the previous interrupt timing, step S
Proceed to ₂₅₀₇ to enter the algorithm processing route, where the variable S
Is set to the value 0. Subsequently, algorithm processing is executed ( _S2508 ).

In this processing, at the previous interrupt timing, each data processed in the operators 1 to 4 and held in each sounding channel area in FIG. 9 is used, and a modulation input for the next operator processing is determined. Is performed. In this process, the contents of the buffer BF or BT are rewritten, and the waveform output value at the interrupt timing is obtained. A specific example of the algorithm processing is shown in the operation flowchart of FIG. In this flow, the same processes as those in FIG. 24 are performed in steps with the same numbers as in FIG. 24. What is different from the case of FIG. 24 is the output part of S _{2601 to} S ₂₆₀₄ . Here, the case of Algorithm 1 and Algorithm 2, the contents of the output O ₁ of the operator 1 processing is maintained as it is to Baffua BF or _{BT (S 2601, S 2602)} . In the case of Algorithm 3, the value which the output O ₃ is added to the output O ₁ is held in Baffua BF or BT (S _2603). Further, in the case of Algorithm 4, a value obtained by adding the outputs O ₂ , O ₃ , and O ₄ to the output O ₁ is held in BF or BT (S ₂₆₀₄ ).

As described above, since the operator process and the algorithm process are alternately executed at every other interrupt timing, the processing load of the sound source processing program per one interrupt timing can be significantly reduced. In this case, it is not necessary to lengthen the interrupt cycle, so that the interruption time of the main operation flowchart of FIG. 4A does not increase due to the interrupt, and the processing load is reduced without affecting the program operation. Can be. Therefore, for example, the interval of taking in the keyboard key executed according to FIG. 4A does not become long, and the response performance as the electronic musical instrument is not affected.

The operation of generating musical sound data for each sound generation channel by software sound source processing based on various sound source systems has been described above.

Function Key Processing Next, when playing an actual electronic musical instrument, the fourth
The specific operation of the function key processing ( _S403 ) in the main operation flowchart of FIG.

In the sound source processing performed for each sounding channel described above, for example, a function key 801 in FIG. 8A connected to an operation panel of an electronic musical instrument via an input port 210 of the MCPU 101 (see FIG. 2) MCPU101 or SCPU102
The data formats (see FIG. 11) of the various sound generation systems described above are set in each sounding channel area (see FIG. 9) on the RAM 206 or 306 (see FIGS. 2 and 3).

FIG. 27 is a diagram showing an example of a partial arrangement of the function keys 801 in FIG. 8 (a). In the figure, a part of the function key 801 is realized as a tone designation switch, and when switches such as “Piano”, “Guitar”,... Is selected, and the guide lamp is turned on. And DPCM / TM system selection switch 2
At 701, it is selected whether the tone of these instrument sounds is generated by the sound source system of the DPCM system or the TM system.

On the other hand, when the "Tuba" switch of group B is pressed, the FM system is used, when the "Base" is pressed, both PCM / TM systems are used, and when the "Trumpet" is pressed, the PCM system is used. A tone color is specified, and a musical tone based on those sound source systems is generated.

FIGS. 28 (a) and (b) show the assignment of the tone generator system to each sounding channel area shown in FIG. 9 on the RAM 206 or 306 when the "piano" and "bass" switches are pressed. An example is shown. In the case of "piano", the DPCM system is assigned to all of the eight-tone polyphonic sounding channels of the MCPU 101 and the SCPU 102, as shown in FIG. 2A, and in the case of "bass", as shown in FIG. The PCM system is assigned to odd-numbered sounding channels, and the TM system is assigned to even-numbered sounding channels. As a result, a musical sound waveform for one sound is obtained as a mixture of the musical sound waveforms generated in the two sounding channels of the PCM system and the TM system. In this case, 4 tones are polyphonic for each CPU, and 8 tones are polyphonic in total for 2 CPUs.

FIG. 29 is a part of the operation flowchart of the function key process of _S403 in the main operation flowchart of FIG. 4A, and is an operation flowchart of the process for the tone color designation switch group of FIG.

First, it is determined whether or not the player has operated the DPCM / TM switch 2701 ( _S2901 ), and if the determination is YES, it is determined whether or not the variable M is zero ( _S2902 ). The variable M is
It is secured in the RAM 206 (FIG. 2) of the MCPU 101 and takes a value of 0 for the DPCM system and a value of 1 for the TM system. If the value of variable M in S ₂₉₀₂ is its determination at 0 YES, the value 1 is set to the variable M (S _2903). This means that the DPCM / TM switch 2701 has been pressed and the TM system has been selected in a state where the DPCM system has been selected. Also, S
_If the value of the variable M is 1 and the determination is NO at ₂₉₀₂ , the value 0 is set to the variable M ( _S2904 ). This is because the DPCM / TM switch 2701 is pressed and the DPCM
It means that the method has changed to the selected state.

Next, it is determined whether or not the tone color of the group A in FIG. 27 is currently designated ( _S2905 ). DPCM / TM switch
Since 2701 is effective only for the tone of Group A,
Tone of A group is designated the determination of S ₂₉₀₅ only if YES, the operation corresponding to the DPCM / TM switch 2701 S ₂₉₀₆ to S ₂₉₀₈ is executed.

In _S2906 , it is determined whether or not the variable M is 0 ( _S2906 ).

If the determination in _S2906 is YES, the DPCM system is selected by the DPCM / TM switch 2701, and the RAMs 206 and 306 (second
Data is set in each sounding channel area in the upper part in the DPCM format shown in FIG. That is, a sound source method No. indicating the DPCM method is set in the head area G (see the column of DPCM in FIG. 11) of each sounding channel area. Subsequently, various parameters corresponding to the currently designated timbre are set in the second and subsequent areas of each sounding channel area ( _S2907 ).

If the determination in _S2906 is NO, the DPCM / TM switch 27
Since the TM method is selected in 01, data is set in each sounding channel area in the TM format shown in FIG. That is, first, a sound source system No. indicating the TM system is set in the head area G of each sounding channel area. Subsequently, various parameters corresponding to the currently designated timbre are set in the second and subsequent areas of each tone generation channel area ( _S2908 ).

The above is the case where the DPCM / TM switch 2701 in FIG. 27 is operated, but the switch is not operated and step S ₂₉₀₁
If the determination in step _S2905 is NO, or if the tone of group A has not been specified and the determination in step _S2905 has been NO, the processing of switch _{S2909 and subsequent} steps is executed.

First, in step _S2909 , it is determined whether or not the tone color switch in FIG. 27 has changed ( _S2909 ).

If the determination is NO, there is no need to perform the processing for the tone color switch, and the function key processing ( _{S403 in} FIG. 4 (a)) is terminated.

If there is a change in the timbre switch and the determination in _S2909 is YES, it is next determined whether or not the timbre of the B group has been designated ( _S2910 ).

If the tone of the B group is designated and the determination in S ₂₉₁₀ is YES, the tone generator data format corresponding to the designated tone is stored in each sounding channel area on the RAMs 206 and 306 (see FIGS. 2 and 3). Is used to set the data. And
A sound source system number indicating each sound source system is set in the head area G (see FIG. 11) of each sound channel area. In addition, various parameters corresponding to the currently designated tone color are set in the second and subsequent areas of each sounding channel area (S ₂₉₁₁ ). For example, when the base switch shown in FIG. 27 is selected, data corresponding to the PCM system is stored in each odd-numbered sound channel region, and data corresponding to the TM system is stored in each even-numbered sound channel region. , Are set respectively.

In addition, the tone switch of the group A is designated and S ₂₉₁₀
If NO is determined, it is determined whether or not the variable M is 1 ( _S2912 ). And if the TM method is currently selected and S
If the determination in ₂₉₁₂ is YES, data is set in each sounding channel area in the TM format shown in FIG. 11 in the same manner as in step _S2908 ( _S2913 ).

If the DPCM method is selected and the determination in _S2912 is NO, data is set in each sounding channel area in the DPCM format shown in FIG. 11 in the same manner as in step _S2907 described above ( S ₂₉₁₄ ).

First Embodiment of Keyboard Key Processing at the Time of Depressing a Key
The specific operation of the keyboard key processing ( _S405 ) in the main operation flowchart of FIG.

First, a first embodiment of keyboard key processing at the time of key depression will be described.

In the first embodiment of the keyboard key processing at the time of key depression,
When the tone of group A in Fig. 27 is specified, the MCP
According to the position of the keyboard when the keyboard key 802 shown in FIG. 8 (a) or 8 (b) connected via the input port 210 (see FIG. 2) of the U101 is depressed, that is, the range of the musical tone, the MCPU 101 Alternatively, the tone generator system set in each sound channel area (see FIG. 9) on the RAM 206 or 306 (see FIGS. 2 and 3) of the SCPU 102 is automatically switched.
In this case, when the key code of the depressed key is in the low tone range of 31 or less, the DPCM method is performed with the key codes 31 and 32 of the keyboard key 801 shown in FIG. In addition, when the key code is in the high range of 32 or more, the TM method is assigned. When the tone color of the group B in FIG. 27 is designated, no special keyboard key processing is executed.

FIG. 30 is a part of the operation flowchart of the keyboard key processing of _S405 in the main operation flowchart of FIG. 4 (a), and the processing at the time of key depression of the keyboard key 801 of FIG. 8 (a) or (b). 5 is a flowchart of the first embodiment.

First, it is determined whether or not the tone color of the group A in FIG. 27 is currently specified ( _S3001 ).

If this determination is NO and the tone color of group B is currently specified, the special processing in FIG. 30 is not performed.

If the determination in _S3001 is YES, and the tone color of Group A is currently specified, it is determined that the key is pressed in the keyboard key loading process in _S404 in the main operation flowchart of FIG. 4A. It is determined whether the key code of the key is 31 or less ( _S3002 ).

If the low tone range 31 or lower is depressed and the determination in S ₃₀₀₂ is YES, it is determined whether or not the variable M is the value 1 (S ₃₀₀₃ ). The variable M is a part of the function key process of _S403 in the main operation flowchart of FIG.
29. As described above, the value of the variable M is set to 0, TM
In the case of the method, the value 1 is taken.

If the determination in _S3003 is YES (M = 1) and the TM system sound source system is currently specified, the sound source system is in the low frequency range.
In order to change to the DPCM system, data is set in the DPCM format shown in FIG. 11 in the sound channel area on the RAM 206 or 306 (see FIGS. 2 and 3) to which the depressed key is assigned. That is, the leading area G of each sounding channel area
(Refer to the DPCM column in FIG. 11.)
o. is set. Then, the second of each sounding channel area
Various parameters corresponding to the currently designated timbre are set in the areas subsequent to the first one ( _S3004 ). Thereafter, the value 1 is set in the flag C ( _S3005 ). The flag C is a variable secured in each sounding channel area on the RAM 206 (see FIG. 2) of the MCPU 101 (see FIG. 11). The flag C is used for key release processing described later with reference to FIG. Used.

On the other hand, in S ₃₀₀₂ , if the treble range of No. 31 or more is depressed and the determination is NO, it is further determined whether or not the variable M is the value 1 (S ₃₀₀₆ ).

If the determination in S ₃₀₀₆ is NO (M = 0) and the sound source system of the DPCM system is currently specified, the key on the RAM 206 or 306 is depressed in order to change to the TM system, which is the sound source system of the high frequency range. Data is set in the TM format of FIG. 11 in the sound channel area to which the key is assigned. That is, a sound source system No. indicating the TM system is set in the head region G (see the column of TM in FIG. 11) of each sounding channel region. Subsequently, various parameters corresponding to the currently designated timbre are set in the second and subsequent areas of each sounding channel area ( _S3007 ). Thereafter, the value 2 is set in the flag C ( _S3005 ).

In the above-described processing, if the determination in step _S3003 is NO and the determination in step _S3006 is YES, no special processing is performed because the desired sound source method has been used originally.

Second Embodiment of Keyboard Key Processing at the Time of Key Depression Next, a second embodiment of the keyboard key processing at the time of key depression will be described.

In the second embodiment of the keyboard key processing at the time of key depression,
When the tone of group A in Fig. 27 is specified, the MCP
8A or 8B, which is connected via the input port 210 of U101 (see FIG. 2), the key pressing speed or velocity when the key 802 is pressed is determined by the following equation.
The RAM 206 or 306 of the CPU 101 or SCPU 102 (FIG. 2 and FIG.
The sound source system set in each sounding channel area (see FIG. 9) is automatically switched. in this case,
MIDI (Musical Instrument Didit
al Interface) The DPCM method is used in the case of a fast key press operation where the velocity value of the key pressed is 64 or more, which is 64 of 1/2 of 127 of the maximum value of the standard. The TM method is assigned to a slow key press operation of 64 or less. When the tone color of the group B in FIG. 27 is designated, no special keyboard key processing is executed.

FIG. 31 is a part of the operation flowchart of the keyboard key processing of _S405 in the main operation flowchart of FIG. 4 (a), and the processing at the time of key depression of the keyboard key 801 of FIG. 8 (a) or (b). 9 is a flowchart of the second embodiment.

First, it is determined whether or not the tone color of the group A in FIG. 27 is currently designated ( _S3101 ).

If this determination is NO and the tone color of group B is currently specified, the special processing in FIG. 30 is not performed.

If the determination in _S3101 is YES and the tone of the group A is currently specified, it is determined that the key is pressed in the keyboard key loading process in _S404 in the main operation flowchart in FIG. 4A. It is determined whether or not the velocity of the key is 64 or more ( _S3102 ). In addition, 64 of this velocity is equivalent to mp (mezzo piano) of MIDI standard.

If the velocity value is 64 or more and the determination in S ₃₁₀₂ is YES, it is determined whether or not the variable M is the value 1 (S ₃₁₀₃ ). The variable M is a part of the function key process of _S403 in the main operation flowchart of FIG.
29. As described above, the value of the variable M is set to 0, TM
In the case of the method, the value 1 is taken.

If the determination in _S3103 is YES (M = 1) and the TM system sound source system is currently specified, in order to change to the DPCM system which is the sound source system at the time of a fast key depression operation, the process shown in FIG. Similar to the processing of _S3004 in the case of the first embodiment, the DPCM format of FIG. 11 is stored in the sound channel area of the RAM 206 or 306 (see FIGS. 2 and 3) to which the depressed key is assigned. Is set ( _S3104 ), and the value 1 is set to the flag C ( _S3105 ).

On the other hand, if the velocity value is smaller than 64 in S ₃₁₀₂ and the determination is NO, it is further determined whether or not the variable M is the value 1 (S ₃₁₀₆ ).

If the determination in _S3106 is NO (M = 0) and the sound source system of the DPCM system is currently specified, the first sound source system in FIG. In the same manner as in the case of the processing of S ₃₀₀₇ in the case of the embodiment, the RAM 206 or 306
Data is set in the TM format of FIG. 11 in the sounding channel area to which the pressed key is assigned (S
₃₁₀₇ ), the value 2 is set to the flag C ( _S3108 ).

In the above-described processing, when the determination in step _S3103 is NO and when the determination in step _S3106 is YES, the desired sound source method is originally used, and no special processing is performed.

Example of Keyboard Key Processing at Key Release Next, an example of keyboard key processing at key release will be described.

According to the first or second embodiment of the keyboard key processing at the time of key depression, the sound source system can be automatically changed by the key range (tone range) and the velocity. Need to be returned to. This is realized by the embodiment of the keyboard key processing at the time of key release described below.

FIG. 32 is a part of the operation flowchart of the keyboard key processing of _S405 in the main operation flowchart of FIG. 4 (a), and the processing at the time of key release of the keyboard key 801 of FIG. 8 (a) or (b). It is a flow of.

First, in the RAM 206 or 306 (see FIGS. 2 and 3) to which the key determined to be "key released" in the keyboard key loading process of _S404 in the main operation flowchart of FIG. 4A is assigned. The value of the flag C set in the sound channel area is determined. Now, the flag C is set to S in FIG.
₃₀₀₅ and S ₃₀₀₈ or set in the processing of S ₃₁₀₅ or S _{3108 in} FIG. 31, the initial value at the time of key pressing is 0,
Value 1 when the sound source system is changed from the system to the DPCM system
Is set, and the value 2 is set when the sound source method is changed from the DPCM method to the TM method. Therefore, if no change is made at the time of key depression, the initial value remains at 0.

Then, in the key release process of FIG. 32, step S
If the value of the flag C is 0 in the determination at ₃₂₀₁ , since the tone generator method has not been changed by the key range or velocity, normal key release processing is performed without performing any special processing.

The determination of S _3201, if the value of the flag C is not determined 1, the sound source mode is changed to DPCM method from the TM mode at the time of key depression. Therefore, in order to return to the TM method, the RAM 206 or 306
(See FIG. 2 or FIG. 3.) Data is set in the TM format of FIG. 11 in the sounding channel area to which the depressed key is assigned. That is, a sound source system No. indicating the TM system is set in the head region G (see the column of TM in FIG. 11) of each sounding channel region. Subsequently, various parameters corresponding to the currently designated timbre are set in the second and subsequent areas of each sounding channel area (S
₃₂₀₂ ).

On the other hand, the determination of S _3201, if the value of the flag C is determined to be 2, the sound source mode is changed to the TM mode from DPCM method at key depression. Therefore, to return to the DPCM method, RAM2
Data is set in the DPCM format of FIG. 11 in the sounding channel area to which the depressed key on 06 or 306 is assigned. That is, the sound source system indicating the DPCM system is provided in the head area G of each sounding channel area (see the column of DPCM in FIG. 11)
No. is set. Subsequently, various parameters corresponding to the currently designated timbre are set in the second and subsequent regions of each sounding channel region ( _S3203 ).

After the above operation, the value of the flag C is returned to 0, the processing in FIG. 32 is terminated, and then a normal key release processing (not shown) is executed.

Aspects of Other Embodiments In the series of embodiments of the present invention described above,
As shown in FIG. 1, two CPUs, MCPU 101 and SCPU 102, share and process different sounding channels, but the number of CPUs may be one, or three or more.

The control ROMs 201 and 301 shown in FIGS.
In addition, if the external memory 116 is constituted by a ROM card, the RO
Various sound source systems can be provided to the user by the M card.

Further, the input port 210 of the MCPU 101 in FIG.
It is possible to connect various operation units in addition to the musical instrument operation unit as shown in the figure, thereby realizing various forms of electronic musical instruments. Further, it is easy to realize as a sound source module that inputs performance information from another electronic musical instrument and performs only sound source processing.

On the other hand, various modes are conceivable in which the tone generator system is assigned to each sounding channel by the function key 801 or the keyboard key 802 in FIG.

Various modulation schemes can be applied other than the FM and TM schemes.

In the modulation method, the present embodiment has been described for the case of four operators, but the number of operators is not limited to this.

〔The invention's effect〕

According to the present invention, a general-purpose processor configuration can be achieved without any need for a dedicated tone generator circuit. As a result, the circuit scale of the entire musical tone waveform generator can be significantly reduced, and even in the case of LSI, the same manufacturing technology as that of a normal microprocessor can be used, and the yield of chips can be improved. Can be greatly reduced. Since the tone signal output means can be constituted by a simple latch circuit, there is almost no increase in manufacturing cost due to the addition of this part.

Further, when it is desired to change the modulation method between, for example, the phase modulation method and the frequency modulation, or when it is desired to change the polyphonic number, it is possible to cope with it only by changing the sound source processing program stored in the program storage means. Development cost can be greatly reduced, and a new sound source system can be provided to the user by, for example, a ROM card or the like.

In this case, the data architecture in which the data link between the performance emotion processing program and the sound source processing program is performed through the musical sound generation data in the data storage means, and the sound source processing program is executed at predetermined time intervals with respect to the performance information processing program. Is realized, the processing for synchronizing the two processors becomes unnecessary, and the program can be greatly simplified. As a result, sound source processing whose processing is complicated, such as a modulation method, can be executed with a sufficient margin.

Furthermore, since the change in the processing time due to the type of modulation method or the selection of the tone generation algorithm in the modulation method can be completely absorbed by the tone signal output means, the tone signal is output to a D / A converter or the like. There is also an effect that a complicated timing control program is not required.

[Brief description of the drawings]

FIG. 1 is an overall configuration diagram of an embodiment according to the present invention, FIG. 2 is an internal configuration diagram of a master CPU, FIG. 3 is an internal configuration diagram of a slave CPU, and FIGS. FIG. 5 is a conceptual diagram showing the relationship between the main operation flowchart and interrupt processing, FIG. 6 (a) is a configuration diagram of a conventional D / A converter unit, FIG. FIG. 7B is a configuration diagram of a D / A converter unit according to the present embodiment, FIG. 7 is a timing chart in D / A conversion, FIG. 8A is a layout diagram of function keys and keyboard keys, FIG. 8 (b) is an explanatory diagram of keyboard keys, FIG. 9 is a diagram showing a storage area for each sounding channel on a RAM, and FIG. 10 is a concept when selecting a sound source processing method for each sounding channel. FIG. 11, FIG. 11 is a configuration diagram of a data format for each sound source method on the RAM, and FIG. 12 is a diagram showing a buffer area on the RAM. 13 is an operation flowchart of sound source processing by the PCM method, FIG. 14 is an operation flow chart of sound source processing by the DPCM method, and FIGS. 15 (a) and (b) show the difference value D and the current address _AF . used interpolated value principle explanatory view of a case of obtaining the X _Q, view the. 16 (a), the operation flow chart of the tone generator processing by FM system (part 1), view the. 16 (b), the sound source processing by the FM system (the FIG. 17 (a) is an operation flowchart of sound source processing (1) by the TM method, and FIG. 17 (b) is an algorithm of sound source processing (1) by the TM method. FIG. 18 is a diagram showing an example of an algorithm in a modulation method. FIG. 19 is an operation flowchart of sound source processing (part 2) by the FM method. FIG. 20 is a sound source processing (TM) by the TM method (part 2). FIG. 21 is a flowchart showing the operation of the modulation method. Operation flow chart of the first improved example, FIG. 22 (a) is an operation flow chart of the operator 1 processing by the FM method according to the first improved example, and 22 (b) is the FM method according to the first improved example. FIG. 23 (a) is an operation flowchart of an operator 1 process by the TM method according to a first improvement example, and FIG. FIG. 24 is a diagram showing an operation algorithm per operator in operator 1 processing by the TM method according to the improved example of FIG. 24; FIG. 24 is an operation flowchart of algorithm processing according to the first improved example; FIG. 26 is an operation flowchart of an algorithm process according to the second improvement example, FIG. 27 is a layout diagram of a part of function keys, and FIGS. 28 (a) and (b) are ,pronunciation FIG. 29 is a diagram showing an example of assigning sound source systems to channels, FIG. 29 is an operation flowchart of function key processing, FIG. 30 is an operation flowchart of a first embodiment of keyboard key processing at the time of key depression, FIG. Is an operation flowchart of the second embodiment of the keyboard key processing at the time of key depression, and FIG. 32 is an operation flowchart of the embodiment of the keyboard key processing at the time of key release. 101: Master CPU, 102: Slave CPU, 103: Address latch for MCPU external memory access, 104: Address latch for SCPU external memory access, 105: Access address conflict avoidance circuit, 106: External Memory selector section, 107 Left D / A converter section, 108 Right D / A converter section, 109 Input port, 110 Output port, 111, 112 Output terminal, 113 Left Output terminal, 114: Write output terminal, 115: External memory data-in terminal, 116: External memory, 201, 301: Control ROM, 202, 302: ROM address decoder, 203: Interrupt control unit, 204, 304: RAM address control unit, 205, 305: ROM address control unit, 206, 306: RAM, 207, 307: Command analysis unit, 208, 308: ALU unit, 209, 309: Multiplication 210, input port, 211, output port, 212, SCPU internal RAM Addressing bus interface, 213: Data bus interface for writing to SCPU, 214: Data bus interface for reading from SCPU, 215: Address bus interface for external memory access, 216: External memory data bus 217: D / A data transfer bus interface, 303: SCPU internal RAM addressing bus interface by MCPU, 310: Write data bus interface from MCPU, 311: Read data bus to MCPU Interface, 312: Address bus interface for external memory access, 313: External memory data bus interface, 601, 602: Latch, 603: D / A converter, 801: Function key, 802: Keyboard Key.

────────────────────────────────────────────────── ─── Continued on front page (72) Inventor Koichiro Taiki 3-2-1, Sakaemachi, Hamura-cho, Nishitama-gun, Tokyo Inside Casio Computer Co., Ltd. Hamura Technical Center (72) Inventor Kazuo Ogura Hamura-cho, Nishitama-gun, Tokyo 3-2-1 Sakaemachi Casio Computer Co., Ltd. Hamura Technology Center (72) Inventor Jun Hosoda 3-2-1 Sakaemachi Hamura-cho, Nishitama-gun, Tokyo Casio Computer Co., Ltd. Hamura Technology Center (58) Field (Int.Cl. ⁷ , DB name) G10H 7/02 G10H 1/02

Claims (2)

(57) [Claims] 1. A program storage means for storing a performance information processing program for processing performance information, a sound source processing program for obtaining a polyphonic tone signal in real time at a predetermined sampling period, and said program storage means. Address control means for controlling an address for reading out a program stored in the memory, decoding means for decoding a program instruction read from the program storage means by the address control means, and necessary for generating a tone signal. Data processing means for storing musical tone generation data, arithmetic processing means, and normally, causing the performance processing means to execute arithmetic processing in accordance with each instruction of the performance processing program decoded by the decoding means. Generating tone generation data necessary for generating a signal and storing the data; At the same time, the control is transferred to the sound source processing program at a predetermined time interval independent of the pitch of the musical tone to be generated, and according to each instruction of the sound source processing program decoded by the decoding means, the data is stored in the data storage means. The arithmetic processing means executes arithmetic processing based on the stored musical sound generation data to generate a polyphonic musical sound signal in real time and asynchronously with the predetermined time interval. Program execution means for transferring control to the control unit; and temporarily holding the tone signal generated asynchronously with the predetermined time interval and the sampling period, and outputting the held tone signal in synchronization with the sampling period. And a musical tone signal output means. . 2. The method according to claim 1, wherein the sampling period corresponds to a predetermined time interval, and the program execution means transfers control to a sound source processing program at the sampling period, and causes the arithmetic processing means to execute arithmetic processing. And that the tone signal output means temporarily holds the tone signal generated asynchronously with the sampling cycle and outputs the tone signal in synchronization with the sampling cycle. The musical sound waveform generator according to claim 1, wherein: