JP2946764B2 - Musical sound wave generator - Google Patents

Description

DETAILED DESCRIPTION OF THE INVENTION

[0001]

BACKGROUND OF THE INVENTION 1. Field of the Invention The present invention relates to a sound source processing system in a musical sound waveform generator, and more particularly, to a musical sound waveform generator in which sound source processing is executed by a plurality of processors.

[0002]

2. Description of the Related Art Various electronic musical instruments having high performance have been realized by the development of digital signal processing technology and LSI processing technology.

[0003] A musical sound waveform generator for an electronic musical instrument requires a large amount of high-speed digital operation. Therefore, conventionally, a dedicated tone generator circuit which realizes an architecture equivalent to a tone generating algorithm based on a required tone generator system by hardware. Was composed by. For this reason, the hardware scale becomes large, and it is difficult to freely change the sound source system. Further, when such a musical tone waveform generator is realized as an electronic musical instrument, processing of the performance information and sound source However, the control for synchronizing both of the processes is complicated, resulting in a great increase in cost in the development thereof.

On the other hand, in recent years, many high-performance microprocessors for performing general-purpose data processing have been realized, and a musical tone waveform generator that performs sound source processing in a software manner using such a microprocessor. Is realized, and in particular, a method of distributing the processing to a plurality of processors is considered.

In such a system, for example, one main processor (master processor) usually executes a performance information processing program to process performance information generated from a keyboard or the like based on a performance operation, and a predetermined time interval. , The processing is shifted to a sound source processing program to generate a musical sound. At this time, the sound source processing program is also executed in parallel by other processors (slave processors). After that, when the execution of the sound source processing program is completed in all the processors, the master processor releases the interrupt and executes the performance information processing. Resume program execution.

In such a system, the master processor and the slave processor each execute a sound source processing program for a plurality of different sound generation channel groups. A method generally considered as a control to be executed is a method of assigning sounds in order from a vacant sound channel.

[0007]

Therefore, there is a possibility that the work amount is uneven among the processors, and for example, the operation amount of executing the program only by the master processor increases. In other words, when sound source processing is executed by software, the interrupt time of some processors becomes longer,
There is a problem that the operation rate of another processor is reduced.

Further, when the interruption time increases, the time for executing the performance information processing program decreases, and it becomes impossible to execute sophisticated and complicated performance information processing or the like to be processed in real time. doing.

It is an object of the present invention to realize efficient sound source processing in each processor when performing sound source processing by a plurality of processors, and to enable advanced performance information processing.

[0010]

According to the present invention, first, data storage means for storing tone control data for each of a plurality of tone generation channels, and a tone generator processing program for obtaining tone signals at predetermined time intervals are executed to execute each of the programs. And a program executing means for generating a tone signal based on the corresponding tone control data in the data storage means for each sounding channel.

[0011] Further, it has the following performance information processing processor. That is, the performance information processing processor executes the performance information processing program for processing the performance information and controls the tone control data in the data storage means of each sound source processing processor. The tone control data of each tone generation channel is controlled so that the channels are used equally and the tone generation is performed. Then, the performance information processing processor causes the program execution means of each sound source processing processor to execute the sound source processing program in parallel at predetermined time intervals.

In the above configuration, the performance information processing processor may be shared by one of the plurality of tone generator processors. In this case, the one sound source processor (master processor) executes the performance information processing program at normal times, and executes interrupt processing at predetermined time intervals to each sound source processor (master processor and slave processor) in parallel. The sound source processing program is executed, and when the execution is completed, the control is returned to the performance information processing program again and executed.

[0013]

According to the present invention, when sound source processing is executed by distributed processing using a plurality of sound source processing processors, the performance information processing processors for executing performance information processing use the sound generation channels of each sound source processing processor equally. The tone control data of each tone generation channel is controlled so that a tone is generated.

Thus, the load on each processor can be made uniform, and the time required for the entire sound source processing per one timing can be shortened. Furthermore, when the sound source processing is realized by an interrupt program for the performance information processing program, the interrupt time is shortened, so that a lot of time can be allocated to the performance information processing program, and sophisticated and complicated performance information processing can be executed in real time. can do.

[0015]

Embodiments of the present invention will be described below with reference to the drawings. <Structure of the present embodiment> FIG. 1 is an overall structural diagram of the present embodiment, which is constituted by one chip except for the external memory 116, in which two CPUs (central processing control units) of a master and a slave are mutually connected. While exchanging information, it performs the sound source processing for musical tone creation.

In FIG. 1, a tone control parameter such as a target value of an envelope value and a tone waveform in a PCM (pulse code modulation) system or a tone difference waveform in a DPCM (differential pulse code modulation) system are first stored in an external memory 116. Etc. are stored.

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
Since both share the waveform data and the like in the external memory 116, conflicts may occur when data is read from the external memory 116 as it is. for that reason,
The MCPU 101 and the SCPU 102 are connected via an MCPU external memory access address latch unit 103 and an SCPU external memory access address latch unit 104, respectively.
By outputting an address signal for external memory access and external memory control data from the output terminals 111 and 112 from the access address conflict avoiding circuit 105, the MCPU
The conflict between the address from the address 101 and the address from the SCPU 102 can be avoided.

The external memory 1 based on the address designation
The data read from 16 is input from the external memory data-in terminal 115 to the external memory selector 106. The external memory selector 106 transfers the read data to the MCPU 1 through the data bus MD based on a control signal from the access address conflict avoidance circuit 105.
01 through the data bus SD and S
Separated into data input to CPU 102,
U 101 and SCPU 102 are input. Thus, data conflict can be avoided.

Then, for each data, MCPU
After the sound source processing is performed by software in 101 and the SCPU 102, all of the sound channels are accumulated, and Le
ft Left output terminal 113 of D / A converter unit 107 and Ri
From the right output terminal 114 of the ght D / A converter unit 108, a left channel left analog output and a right channel right analog output are output as tone signals, respectively.

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

In FIG. 1, a control ROM 201 includes:
A tone control program described later is stored therein.
The OM address control unit 205 sequentially outputs the program word (instruction) at the designated address via the ROM address decoder 202. Specifically, the word length of each program word is, for example, 28 bits, and the next address in which a part of the program word is input to the ROM address control unit 205 as the lower part (intra-page address) of the address to be read next System.

The command analysis unit 207 includes a control ROM 2
It analyzes the operation code of the instruction output from 01 and sends a control signal to each part of the circuit to execute the specified operation.

The RAM address control unit 204 includes a control R
When the operand of the instruction from the OM 201 specifies a register, the address of the corresponding register in the RAM 206 is specified. 19 and 20 are stored in the RAM 206.
In addition to storing various musical tone control data described later for eight tone generation channels, various buffers and the like described later as FIGS. 16 and 21 are stored and used for sound source processing described later.

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

The interrupt control unit 203, based on an internal hard timer (not shown), at regular time intervals,
A reset release signal A, a ROM address control unit 205 and each D / A converter unit 107 in FIG.
An interrupt signal is supplied to 108.

In addition to the above configuration, the MCPU 101 shown in FIG.
Are provided with interfaces for the following various buses.

That is, in order to access the external memory 116, an interface 215 of an address bus MA for specifying an address of the memory, and the accessed data is transferred to the MCPU 101 via the external memory selector 106.
An interface 216 of a data bus MD for transmitting and receiving data to and from the bus M which specifies an address of a RAM inside the SCPU 102 in order to transmit and receive data to and from the SCPU 102.
a, the MCPU 101 is the SCPU 10
Interface 213 of the data bus Dout for writing data to the data bus 2 and interface 2 of the data bus Din for the MCPU 101 to read data from the SCPU 102.
14, a D / A data transfer bus interface 217 for transferring the final output waveform to the Left D / A converter unit 107, the Right D / A converter unit 108, and an external switch unit or keyboard unit (FIG. 7, There are input / output ports 210 and 211 for exchanging data with the like (see FIG. 8).

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

The SCPU 102 only performs the sound source processing in response to the processing start signal from the MCPU 101.
2, an input / output port for exchanging data with external circuits corresponding to 210 and 211 in FIG. 2, a Left D / A converter 107 corresponding to 217 in FIG. 2, and a Right D / A There is no interface for outputting a tone signal to the converter unit 108. Other 30
The circuits 1, 302, 304 to 309 correspond to 20 in FIG.
1, 202, and 204 to 209 have the same functions. Also, each interface 303, 310 to 313
Are provided to face each of 212 to 216 in FIG. 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 stored in the RAM3.
06. By this, for example, SCPU
The accumulated waveform data for up to eight sounding channels generated at 102 and held in the RAM 306 is stored in the data bus D.
It is output to MCPU 101 via IN. This will be described later.

In addition to the configuration shown above, in this embodiment, the input port 210 of the MCPU 101 is
Are connected to a function key 701 and a keyboard key 702 as shown in FIG. These parts constitute a substantial musical instrument operation unit.

Next, FIG. 5 shows the D / D of Left and Right in FIG.
This shows the internal configuration of A-converters 107 and 108 (the contents of both converters are the same). One sample data of a musical tone created by tone generator processing is input to a latch 401 via a data bus. Then, M is input to the clock input of the latch 401.
When a sound source processing end signal is input from the command analysis unit 207 (FIG. 2) of the CPU 101, the tone data for one sample on the data bus is latched by the latch 401.

Here, the time required for the sound source processing is as follows.
It depends on the software for sound source processing. for that reason,
When the sound source systems are different, the sound source processing ends, and the timing at which the tone data is latched in the latch 401 is not constant. Therefore, as shown in FIG. 4, the output of the latch 401 cannot be directly input to the D / A converter 402.

Therefore, in this embodiment, as shown in FIG. 5, the output of the latch 401 is further latched by the latch 501, and the interrupt signal shown in FIG.
The tone signal is latched by the latch 501 and is output to the D / A converter 402 at regular intervals as shown in FIG.

As described above, since two latches are used to absorb a change in processing time due to the sound source system, the musical sound data is stored in D.
A complicated timing control program for outputting to the / A converter is no longer necessary. <Overall operation of this embodiment>

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

In this embodiment, basically, the MCPU 101 operates mainly, and as shown in the main flowchart of FIG. 9, a series of processes of S902 to S910 are repeatedly performed. The actual sound source processing is performed by interrupt processing. Specifically, the MCPU 101 and the SCPU 102 are interrupted at certain intervals,
Based on this, each CPU performs sound source processing for generating a maximum of eight channels of sound. In this case, each C
A characteristic feature is that an equal number of tone generation channels are allocated to PUs, thereby shortening the time required for interrupt processing. When the processing is completed, musical tone waveforms for a maximum of 16 channels in total, ie, up to 8 channels of each CPU, are added, and the left D / A converter unit 107 and Righ
t Output from the 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 general overall operation of the present embodiment. Next, details of the overall operation of the present embodiment will be described with reference to FIGS.
This will be described with reference to FIG.

S902 to S in the main flowchart of FIG.
When an interrupt is received from the interrupt control unit 203 while the process of 910 is repeatedly executed, the MCP of FIG.
The two processes of the U interrupt process and the SCPU interrupt process of FIG. 11 are activated at the same time. 10 and FIG.
No. 1 “sound source processing” is shown in FIG. 15 described later.

The main flowchart of FIG. 9 shows the flow of processing other than the sound source processing, which is executed in the MCPU 101 in a state where no interrupt is issued from the interrupt control unit 203.

First, the power is turned on, and the R
Initial settings such as the contents of the AM 206 are performed (S901).
Here, the ME on the RAM 206 particularly relevant to the present invention is described.
Data of C, SEC, HSS1 to HSS8 and S are respectively MEC = 8, SEC = 8, HSS1 to H
SS8 = 0 and S = 1 are initialized.

Next, a function key connected to the outside of the MCPU 101, for example, a tone switch shown in FIG. 47 is scanned (S902), and the state of each switch is input to the input port 210.
From the key buffer area in the RAM 206. As a result of the scanning, the function key whose state has changed is identified, and the processing of the corresponding function is performed (S903). 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 state of the keyboard key being depressed is fetched in the same manner as in the case of the above-mentioned function key (S904).
A key assignment process is performed by identifying the changed key (key pressed or released) (S905). In this process, data of a musical tone to be generated based on a depressed key is assigned to a tone generation channel, and conversely, a tone generation channel corresponding to a released key is released. The processing of this part is a characteristic of the present embodiment, and will be described later with reference to FIGS. In addition,
The assignment process at the time of key-on, which is a part of the process executed here, is only a process of assigning a key pressed key to a sounding channel, and actually sets pitch data based on a key pressed key to an assigned sounding channel. The operation of issuing a sounding instruction is executed in a sounding process (S909) described later.

Next, when a demonstration performance key is pressed with the function key 701 (see FIG. 7), demonstration performance data (sequencer data) is sequentially read from the external memory unit 116, and
Key assignment processing similar to that in S905 (see FIGS. 13 and 14 described later) and the like are performed (S906). When the rhythm start key is pressed, the rhythm data is sequentially read from the external memory 116, and a key assignment process similar to S905 (see FIGS. 13 and 14 described later) and the like are performed (S907).

Thereafter, a timer process described below is performed (S908). That is, the time value of the time data that has been incremented is determined in an interrupt timer process (S1012) described later, and the sequencer data for time control that is sequentially read for demonstration performance control or the time control for time control that is read for rhythm performance control are determined. By comparing the rhythm data with the rhythm data, time control for performing the demonstration performance of S906 or the rhythm performance of S907 is performed.

Further, in the sound generation process S909, pitch data is set to a sound channel to be assigned and to start sound generation in each of the above steps S905 to S907, and the pitch data of the sound channel being sounded is set according to a preset envelope. Pitch envelope processing or the like is performed in which the pitch is changed over time to add a change to the pitch of the tone being generated.

Further, a one-round flow preparation process is executed (S910). Here, the process of changing the state of the sound channel to which the pitch data is newly set in step S909 during sound generation, and changing the state of the sound channel released in steps S905 to S907 to mute, etc., are performed. Done. Whether each sounding channel is sounding or mute is determined by whether the variables HSS1 to HSS8 corresponding to each sounding channel stored in the RAM 206 (FIG. 2) are 1 or 0.

Next, the MCPU interrupt processing of FIG. 10 will be described.

The interrupt control unit 203 of the MCPU 101
Therefore, when the MCPU 101 is interrupted, the processing of the main flowchart of FIG. 9 is interrupted, and the execution of the MCPU interrupt processing of FIG. 10 is started. In this case, MCPU
In the interrupt processing program, a register or the like to which data is written in the main flow program of FIG. 9 is controlled so that the contents are not rewritten. This eliminates the need to save and restore the registers at the start and end of the normal interrupt processing, and the transition between the processing in the main flowchart of FIG. 9 and the MCPU interrupt processing is performed quickly.

First, in the MCPU interrupt processing, MC
The content of the variable S stored on the RAM 206 of the PU 101 is incremented by 1 (S1010). Since this variable S is a variable used in connection with the algorithm of FIG. 43, it will be described later in the description of the algorithm.

Subsequently, sound source processing is started in the MCPU interrupt processing (S1011). This sound source processing is shown in FIG. 15 described later.

At the same time as the above operation, the SCPU reset release signal A (see FIG. 1) is output from the interrupt control unit 203 of the MCPU 101 to the ROM address control unit 305 of the SCPU 102. Execution of the interrupt processing is started.

At approximately the same time as the sound source process in the MCPU interrupt process (S1011), the sound source process in the SCPU interrupt process is started (S1101). As described above, when the MCPU 101 and the SCPU 102 execute the sound source processing in parallel, the processing speed of the sound source processing is approximately doubled as compared with the case where the sound source processing is executed by one CPU.

Subsequently, after the interrupt timer process of S1012, the MCPU 101 waits for an end signal of the SCPU interrupt process from the SCPU 102 (S10).
13). In the interrupt timer process, the value of time data (not shown) on the RAM 206 (FIG. 2) is incremented by utilizing the fact that the interrupt process of FIG. 10 is executed at a constant sampling cycle. 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 S908 of the main flow of FIG. 9 as described above.

When the sound source processing of step S1101 in the SCPU interrupt processing of FIG.
The SCPU processing end signal B (see FIG. 1) is input from the command analysis unit 307 to the ROM address control unit 205 of the MCPU 101. Thus, the determination in step S1013 in the MCPU interrupt processing in FIG. 10 becomes YES.

As a result, the waveform data generated by the SCPU 102 via the data bus Din of FIG.
The data is read into the AM 206 (S1014). In this case, since the waveform data is stored in a predetermined buffer area (buffer B described later in FIG. 21) on the RAM 306 of the SCPU 102, the command analysis unit 207 of the MCPU 101
U The waveform data is read by designating the buffer address to the RAM address control unit 304 via the internal address designation bus Ma.

Then, in S1014 ', the contents of the buffer area are stored in the Left D / A converter 107 and the Right D / A converter 107.
It is latched by the latch 401 (see FIG. 5) of the / A converter section 108.

Next, FIG. 12 is a flow chart conceptually showing the relationship between the processes in the flowcharts of FIGS. 9 and 10 described above, and shows how the MCPU 101 and the SCPU 102 perform sound source processing in a shared manner. Is shown.

First, a certain process A (hereinafter, processes B, C,...)
.. And F are the same) (S1201). This processing corresponds to, for example, "function key processing" or "keyboard key processing" in the main flowchart of FIG. Then M
CPU interrupt processing and SCPU interrupt processing are entered, and at the same time, sound source processing by the MCPU 101 and the SCPU 102 is started (S1202, S1203). And SCPU 10
At the end of the SCPU interrupt processing in 2, an SCPU processing end signal B is input to the MCPU 101. 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 the Left D / A converter The signal is output to the unit 107 and the Right D / A converter unit 108. Then, the process returns to the process B following the process A in the main flowchart.

The above operation is repeated while sound source processing is performed for all sound channels (sound channels being executed by the MCPU 101 and the SCPU 102) (S
1204 to S1216). This repetition processing is continued while the musical sound is being generated. <Detailed operation of key assignment processing and sound source processing>

Next, FIG. 1 executed in steps S905, S906 and S907 of the main flowchart of FIG.
3. The key assignment process of FIG.
The operation of the sound source processing of FIG. 15 executed in step S1011 of the CPU interrupt processing or step S1101 of the SCPU interrupt processing will be described in detail. These processes are
This is a process particularly related to the present invention.

As described above, the MCPU 101 and the SCPU 1
In No. 02, the processing speed of the sound source processing is increased by executing the sound source processing in parallel with a maximum of eight sounding channels each based on the interrupt processing of FIGS. 10 and 11. In order to thoroughly increase the speed, in the present embodiment, both C and C channels are set so that the tone generation channel on which the sound source processing is to be performed is not biased toward either the MCPU 101 or the SCPU 102.
Assign sound channels to PUs equally,
In the sound source processing executed in each interrupt processing, control is performed so that sound source processing is not executed for sound channels to which no assignment has been made, thereby making it possible to shorten the interrupt processing time. As a result, the loop processing of steps S902 to S910 in the main flowchart of FIG. 9 is performed at high speed, and various key scanning processes can be performed at high speed, so that the response performance as a musical instrument can be improved. it can.

First, FIGS. 13 and 14 are operation flowcharts of the key assignment process executed in steps S905, S906 and S907 of the main flowchart of FIG. First, it is determined whether the key assignment process to be performed is a process for sound generation or a process for silence (S1301). This is a process for determining whether a key-on or a key-off has been instructed.

In the case of sound generation processing, it is next determined whether or not both the contents of the variables MEC and SEC are 0 (S1302). Variables MEC and SEC are stored in the RAM 206 (FIG. 2) in the MCPU 101 as shown in FIG. The variable MEC indicates the number of empty channels available for sound generation processing by the MCPU 101, and the variable SEC indicates
It indicates the number of free channels available for sound processing in the SCPU 102. Here, the maximum number of channels that can be processed by the MCPU 101 and the SCPU 102 is eight each. For this purpose, the values of the variables MEC and SEC are both initially set to 8 as the maximum number of channels in step S901 of the main flowchart of FIG. 9 executed immediately after the power is turned on. If it is determined in step S1302 in FIG. 13 that the contents of the variables MEC and SEC are both 0, no more sound can be generated, and the key assignment process is terminated without performing key assignment. Steps S902 to S910 are repeatedly executed until any sounding channel is released.

If the determination in step S1302 is NO, it is determined whether the value of the variable MEC is equal to or greater than the value of SEC (S1303). If this determination is YES, the number of empty channels of the MCPU 101 is larger, so that an empty channel of the MCPU 101 is secured (S1304). Then, the content of the variable MEC is decremented by 1 (S1305), and the key assignment process ends.

On the other hand, if the determination in step S1303 is NO,
Since the number of empty channels of the SCPU 102 is larger, the SCPU 1
02 is reserved (S1306). Then, the content of the variable SEC is decremented by 1 (S1307), and the key assignment process ends.

As described above, the MCPU 101 and the SCP
Assignment of sound channels in U 102 is performed equally.

On the other hand, if it is determined in step S1301 that the key assignment process to be performed is a process for silencing, it is determined whether a tone to be silenced is assigned to the tone generation channel of the MCPU 101. (S
1308). This information is stored in the RAM 206 (FIG. 2).

If the tone to be muted has been assigned to the MCPU 101 and the determination in step S1308 is YES, the tone generation channel of the MCPU 101 to which the tone is assigned is canceled and made an empty channel (S1309). Then, the content of the variable MEC is incremented by 1 (S1310), and the key assignment process ends.

Conversely, if the tone to be silenced is assigned to the SCPU 102 and the determination in step S1308 is NO,
The tone generation channel of the SCPU 102 to which the tone is assigned is canceled and made an empty channel (S1311).
Then, the content of the variable SEC is incremented by 1 (S1312), and the key assignment process ends.

As will be described later with reference to FIG. 49 and the like, when different sound source systems are assigned to two sounding channels to generate a musical tone waveform for one sound, the key assignment process of FIGS. The process is executed in pairs of two sounding channels, and the change of the variables in steps S1305, S1307, S1310 and S1312 is -2 or +2 each. Similarly, when one sound is generated in n channels, the value is -n or + n each.

As described above, the MCPU 101 or the SCPU
The process of setting pitch data for the tone generation channel secured at 102 based on the note number actually keyed on is executed in step S909 of the main flowchart of FIG. 9 as described above.

In the flow one round preparation process of the subsequent step S910, the state of the sounding channel secured by the above-described key assignment process is changed during sounding, and the state of the released sounding channel is changed to mute. Can be Therefore, the RAM 206 of the MCPU 101 and the R of the SCPU 102
The AM 306 stores variables HSS1 to HSS8 respectively corresponding to the first to eighth sounding channels, and determines whether the corresponding sounding channel is mute or sounding according to whether the value is 0 or 1. Represents The values of these variables are initially set to 0, which indicates that the sound is being muted, in step S901 of the main flowchart of FIG. 9 executed immediately after the power is turned on. And MCPU 101 or S
When the CPU 102 secures a sounding channel by the key assignment process described above, the variable HSSi (1) on the RAM 206 or 306 corresponding to the sounding channel is obtained.
≤ i ≤ 8) is changed from 0 to 1, and conversely, from 1 to 0 when the sound channel is released.

Next, FIG. 15 is a common operation flowchart of the sound source processing executed in step S1011 of the MCPU interrupt processing or step S1101 of the SCPU interrupt processing.

First, an area for adding waveform data (described later with reference to FIG. 21 and the like) in the RAM 206 or RAM 306 is cleared (S1501).

Next, each of the MCPU 101 and the SCPU 102 determines whether the value of the variable HSS1 on the RAM 206 or 306 corresponding to the first sounding channel is 0 or 1 for the first sounding channel ( S15
02) Only when the value is 1, sound source processing is performed for the sounding channel (S1503).

Steps S1504 and S1505, S1506 and S1507, S1508 and S1509, S1510 and S1511, and S1512 and S
1513, S1514 and S1515 and S1516 and S1517, 2
Similar processing is performed for each of the eighth to eighth sounding channels.

When the sound source processing for the eighth channel is completed, waveform data obtained by adding the data for the eight channels to a predetermined buffer area B (described later with reference to FIG. 21) is obtained. Details of these processes will be described later. As can be seen from the above operation, in the sound source processing executed in each interrupt processing, the sound source processing is not executed for the sound-generating channel that is not assigned and is in the mute state. The execution in conjunction with the key assignment process realizes a reduction in the interrupt processing time. <Data structure in sound source processing>

Next, S1011 in FIG. 10 and S1101 in FIG.
A description will be given of a specific example of the sound source processing executed by the above.

In this embodiment, the MCPU 101 and the SCPU 10
As described above, the two CPUs 2 share the sound source processing for each of up to eight channels. As shown in FIG. 17, the MCPU 101, SCPU
It is set in an area for each sounding channel in the RAMs 206 and 306 of the CPU 102.

As shown in FIG. 21, each buffer of S, BF, BT, B, and M is secured in this RAM.

In this case, each sound source system can be set in each sound channel region of FIG. 17 by an operation described in detail later, as conceptually shown in FIG. 18, and when the sound source system is set, Data is set in each area of each sounding channel in FIG. 17 in a data format of each sound source system as shown in FIG. 19 and 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 shown in FIGS. 19 and 20, HSS is a variable indicating whether each sound channel is muting or sounding as described above. G is a sound source system number which is a number for identifying the sound source system.
It is. The next A indicates an address designated when waveform data is read during sound source processing. AI, A1 and A2 are integer parts of the current address, and the waveform data of the external memory 116 (FIG. 1) is stored. Corresponds directly to the address. A F is a decimal part of the current address, and is used for interpolation of the waveform data read from the external memory 116.

The following AE indicates an end address, and AL indicates a loop address. Further, PI, P1 and P2 represent integer parts of the pitch data, and PF represents a decimal part of the pitch data. For example, PI = 1, PF = 0 is the pitch of the original sound, PI = 2, PF = 0 is the pitch one octave higher, and PI = 0, PF = 0.5 is the pitch of one octave lower. Represents the pitch, respectively.

The next XP uses the previous sample data as XN
Represents the storage of the next sample data (described later). 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 tone generator system.

As described above, data as shown in FIGS. 19 and 20 is stored in each RA of the MCPU 101 and the SCPU 102.
When the sound source system described later is determined in M206 and M306, data is set in the format of FIGS. 19 and 20 for each channel shown in FIG.

Hereinafter, the sound source processing of each sound source system executed using such a data structure will be sequentially described. Note that these sound source processes are performed by the MCPU 101 or the SCPU 10
The second command analysis unit 207 or 307 is a control ROM
This is realized by interpreting and executing a sound source processing program stored in 201 or 301. Hereinafter, it is assumed that processing is performed on this premise unless otherwise specified.

First, in the flowchart of FIG.
Upon entering each sound source processing for each channel (S1502 etc.), RA
The sound source system No. of the data of the data format (Table 1) shown in FIGS. 19 and 20 stored in the corresponding sound generation channels of M206 and 306 is determined, whereby the sound source process of any of the sound source systems described below is determined. Is performed. <Sound source processing by PCM method>

When the sound source system number indicates the PCM system, the sound source process according to the PCM system shown in the operation flowchart of FIG. 22 is executed. Each variable in the flow is stored in the RAM 20 of the MCPU 101 or the SCPU 102.
6 and 306 stored in any one of the tone generation channel areas in FIG. 17 in the PCM format of Table 1 in FIGS. 19 and 20.

FIG. 25 shows addresses (AI, AF) of the addresses at which the PCM waveform data stored in the external memory 116 (FIG. 1) are stored at the current processing target. And

First, the pitch data (P
(I, PF) is added (S2201). The pitch data corresponds to the type of the key pressed by the keyboard key 702 or the like in FIG.

Then, the integer part AI of the added address
It is determined whether or not has changed (S2202). Judgment is NO
Then, using the difference value D, which is the difference between the respective sample data XN and XP at the address (AI + 1) and AI in FIG. 25, an interpolation process corresponding to the decimal part AF of the address is performed by an arithmetic operation of D.times.AF. The data value O is calculated (S2207). The difference value D has been obtained by the sound source processing at the interrupt timing before this time (see S2206 described later).

Then, the sample data XP corresponding to the integer part AI of the address is added to the interpolation data value O,
New sample data O (corresponding to XQ in FIG. 25) corresponding to the current address (AI, AF) is obtained (S22).
08).

Thereafter, the sample data is multiplied by an envelope value E (S2209), and the content of the obtained O is stored in the MCP.
It is added to the waveform data buffer B (see FIG. 21) in the RAM 206 or 306 of the U 101 or the SCPU 102 (S2210).

Thereafter, returning to the main flow of FIG. 9, an interrupt is applied in the next sampling period, and the operation flowchart of the sound source processing of FIG. 22 is executed again, and the pitch data (PI, PF) is stored in the current address (AI, AF). ) Is added (S2201).

The above operation is repeated until the integer part AI of the address changes (S2202).

During this time, the sample data XP and the difference value D
Is not updated, only the interpolation data O is updated according to the address AF, and the sample data XQ is obtained each time.

Next, at S2201, the current address (AI, AF)
When the integer part AI of the current address changes as a result of adding the pitch data (PI, PF) to the address (S2202), it is determined whether or not the address AI has reached or exceeded the end address AE (S2203). .

If the determination is YES, the next loop processing is performed. That is, the address exceeding the end address AE (AI-AE) is added to the loop address AL, and the resulting integer part A of the new current address is obtained.
Loop playback is started from I (S2204). The end address AE is an address on the external memory 116 (FIG. 1) where the last waveform sample data of the PCM waveform data is stored. The loop address AL is the address of the position where the player wants to repeat the output of the waveform.
By the above operation, a well-known loop process is realized by the PCM method.

If the determination in S2203 is NO, the process in step S2204 is not executed. Next, the sample data is updated. Here, the external memory 116 (FIG. 1)
), The sample data corresponding to the newly updated current address AI and the immediately preceding address (AI-1) are read out as XN and XP, respectively (S22).
05).

Furthermore, the difference value up to now is the same as the updated X
The difference value D between N and XP is updated (S2206).

The subsequent operation is as described above.

As described above, P for one sounding channel is obtained.
Waveform data according to the CM method is generated. <Sound source processing by DPCM method>

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

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

In the figure, external memory 116 (FIG. 1)
Of the sample data XP corresponding to the address AI of the address (AI-
This is a value obtained from a difference value from the sample data corresponding to 1). Since the next difference value D is written in the address AI of the external memory 116 (FIG. 1), the sample data of the next address is obtained by XP + D, and this is replaced with new sample data XP.

In this case, assuming that the current address is AF as shown in FIG. 26, the sample data corresponding to the current address AF is obtained by XP + D × AF. Thus, DP
In the CM system, the difference value D between the current address and the sample data corresponding to the next address is stored in the external memory 116.
(FIG. 1), and is added to the current sample data to determine the next sample data.
Waveform data is created sequentially.

When such a DPCM system is employed, when quantizing a waveform such as a voice or a musical tone in which the difference between adjacent samples is generally small, the number of bits is much smaller than that of the ordinary PCM system. Can be used to perform quantization.

The operation of the above DPCM system will be described with reference to the operation flowcharts of FIGS. Each variable in the flow is stored in the RAM 206 of the MCPU 101 or the SCPU 1
19 in the DPCM format of Table 1 in FIG. 19 stored in any one of the tone generation channel areas in FIG.

FIG. 26 shows addresses (AI, AF) of the addresses on the external memory 116 (FIG. 1) where the DPCM difference waveform data is stored, where the data to be processed is stored. ).

First, pitch data (PI, PF) is added to the current address (AI, AF) (S2301).

Then, the integer part AI of the added address
It is determined whether or not has changed (S2302). Judgment is NO
Then, using the difference value D at the address AI in FIG.
An interpolation data value O corresponding to F is calculated (S2314).
The difference value D is obtained by the sound source processing at the interrupt timing before this time (S2306 described later).
And S2310).

Next, sample data XP corresponding to the integer part AI of the address is added to the interpolation data value O, and new sample data O (corresponding to XQ in FIG. 26) corresponding to the current address (AI, AF) is obtained. Obtained (S2315).

Thereafter, the sample data is multiplied by an envelope value E (S2316), and the content of the obtained O is stored in the MCP.
RAM 206 of U 101 or RAM 30 of SCPU 102
6 is added to the waveform data buffer B (see FIG. 21) (S2317).

Thereafter, returning to the main flow of FIG. 9, an interrupt is applied in the next sampling cycle, and the operation flowchart of the sound source processing of FIGS. 23 and 24 is executed again.
Pitch data (PI, PF) at current address (AI, AF)
Is added (S2301).

The above operation is repeated until a change occurs in the integer part AI 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, and new sample data XQ is obtained each time.

Next, in S2301, the current address (AI, AF)
If the integer part AI of the current address changes as a result of adding the pitch data (PI, PF) to the current address (S2302), it is determined whether the address AI has reached or exceeded the end address AE (S2303). .

If the judgment is NO, the following S2304 to S2307
, The sample data corresponding to the integer part AI of the current address is calculated. That is, first, the value before the change of the integer part AI of the current address is stored in a variable called old AI (see the column of DPCM in Table 1 in FIG. 19). This value is stored by repeating the processing of S2306 or S2313 described later. This old A
While the value of I is successively incremented in S2306,
External memory 116 indicated by old AI at 07 (FIG. 1)
The above difference waveform data is read out as D, and is sequentially accumulated in step S2305 to sample data XP. Then, when the value of the old AI becomes equal to the integer part AI of the changed current address, the value of the sample data XP becomes a value corresponding to the integer part AI of the changed current address.

Thus, the integer part A of the current address
When the sample data XP corresponding to I is obtained, the determination in S2304 is YES, and the above-described interpolation value calculation processing (S231)
Go to 4).

The above sound source processing is repeated at each interrupt timing, and if the determination in S2303 changes to YES, the next loop processing is started. First, the end address AE
Are added to the loop address AL, and the obtained address is used as the integer part AI of the new current address (S2308).

Thereafter, the operation of accumulating the difference value D is repeated several times depending on how much the address has advanced from the loop address AL, whereby the sample data XP corresponding to the integer part AI of the new current address is calculated. Is done. That is, first, as an initial setting, the sample data X
P is the value of the sample data XPL at the preset loop address AL (see the DPCM column in Table 1 of FIG. 19), and the old AI is the loop address AL.
(S2309). Then, the following S2310 to S23
The process of 13 is repeated. That is, the value of the old AI is S23
While being sequentially incremented at 13, the difference waveform data on the external memory 116 (FIG. 1) designated by the old AI is read as D at S2310, and is sequentially accumulated at S2312 to the sample data XP. Then, when the value of the old AI becomes equal to the integer part AI of the new current address,
The value of the sample data XP becomes a value corresponding to the integer part AI of the new current address after the loop processing.

When the sample data XP corresponding to the new integer part AI of the current address is obtained in this manner, S
When the determination at 2311 is YES, the process proceeds to the above-described interpolation value calculation process (S2314).

As described above, D for one sounding channel is obtained.
Waveform data according to the PCM method is generated. <First sound source processing by FM modulation method>

Next, the first sound source processing based on the FM modulation method will be described. In the case of the FM modulation method, usually, FIG.
As shown in OP1 to OP4, hardware or software modules having the same function called operators are used, and they are connected as shown in algorithms 1 to 4 in FIGS. 31 (a) to 31 (d). Musical sounds are generated by being connected to each other by rules. In this embodiment, the FM modulation method is realized by software processing.

Next, the operation of the first embodiment in the case where sound source processing is performed by two operators will be described with reference to the operation flowchart of FIG. The processing algorithm is shown in FIG. Each variable in the flow is the MCPU 101 or SCPU
Each of the data in the FM format of Table 1 in FIG. 20 is stored in one of the sound channel areas in FIG. 17 on the RAMs 206 and 306 of the CPU 102.

First, the operator 2 which is a modulator
The processing of (OP2) is performed. For pitch processing, PC
Since interpolation is not performed unlike the M method, the integer address A
There are only two. 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 is stored in the current address A2.
2 is added (S2701). Next, this address A2
And the feedback output FO2 is added as the modulation input,
A new address AM2 is obtained (S2702). The feedback output FO2 is obtained by executing the process of S2705 described later at the previous interrupt timing.

Further, a sine wave value corresponding to the address AM2 (phase) is calculated. In practice, the external memory 116
The sine wave data is stored in FIG. 1 (FIG. 1), and is obtained by looking up the sine wave data at the address AM2 (S2703).

Subsequently, the sine wave data is multiplied by an envelope value E2 to obtain an output O2 (S2704).

Thereafter, a feedback signal is output to this output O2.
The level FL2 is multiplied to obtain a feedback output FO2 (S2705). In the case of the present embodiment, the output FO2 is used by the operator 2 (OP2) at the next interrupt timing.
Input to

Further, the modulation level M is applied to O2.
L2 is multiplied to obtain a modulation output MO2 (S2706). This modulation output M02 becomes a modulation input to the operator 1 (OP1).

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

First, the pitch data P1 is added to the current address A1 of the operator 1 (S2707), and the above-mentioned modulation output MO2 is added to this value to obtain a new address AM1 (S2708).

Next, the value of the sine wave corresponding to the address AM1 (phase) is read from the external memory 116 (FIG. 1) (S2709), and is multiplied by the envelope value E1 to obtain a musical sound waveform output O1. (S2710).

This is added to the buffer B (see FIG. 21) in the RAM 206 (FIG. 2) or 306 (FIG. 3) (S2711), and the FM modulation processing for one tone generation channel ends. <First sound source processing by TM modulation method>

Next, the first sound source processing by the TM modulation method will be described.

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

The above-mentioned FM modulation method is as follows.

[0139] The operation formula is basically based on the following equation: e = A · sin {ωct + I (t) · sinωmt}. Here, ωct is the carrier wave phase angle (carrier signal), sin ωmt is the modulation wave phase angle (modulation signal),
And I (t) are the modulation indices.

On the other hand, the phase modulation method referred to as the TM modulation method in the present embodiment is based on an arithmetic expression of e = A · fT ｛fc (t) + I (t) · sinωmt｝. Here, fT (t) is a triangular wave function, and is defined by the following function for each phase angle region (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) ・ ・ ・ (Area: 3π / 2 ≦ ω ≦ 2π)

Further, fc is called a modified sine wave, and is obtained by accessing the external memory 116 (FIG. 1) in which different sine waveform data is stored for each region of each phase angle with the carrier phase angle ωct. This is a carrier signal generation function. Fc for each phase angle region is defined as follows.

Fc (t) = (π / 2) sinωct (region: 0 ≦ ωct ≦ π / 2) fc (t) = π− (π / 2) sinωct (region: π ≦ ωct) ≦ 3π / 2) fc (t) = 2π + (π / 2) sinωct (region: 3π / 2 ≦ ωct ≦ 2π)

In the TM modulation method, the function fc (t) as described above is used.
The modulation signal sin ωmt to the carrier signal generated by
The above-described triangular wave function is modulated by an addition signal obtained by addition at a ratio indicated by I (t). This gives the modulation index I
If the value of (t) is 0, a sine wave can be generated, and I
By increasing the value of (t), a very deeply modulated waveform can be generated. Here, various signals can be used in place of the modulation signal sin ωmt, and as described below, the own operator output at the previous calculation is fed back at a constant feedback level, and the output of another operator is input. Or can be.

The sound source processing based on the TM modulation method based on such a principle will be described with reference to the operation flowchart of FIG. Also in this case, similar to the case of the FM modulation scheme in FIG. 27,
This is an example in which the sound source processing is performed by two operators, and the processing algorithm is shown in FIG. Each variable in the flow is stored in the RAM 20 of the MCPU 101 or the SCPU 102.
The data in the TM format of Table 1 in FIG. 20 stored in any one of the sound channel areas in FIG.

First, the operator 2 which is a modulator
The processing of (OP2) is performed. For pitch processing, PC
Since interpolation is not performed unlike the M method, the integer address A
There are only two.

First, the pitch data P is stored in the current address A2.
2 is added (S2901). Next, a modified sine wave corresponding to the address A2 (phase) is read from the external memory 116 (FIG. 1) by the modified sine transform fc, and a carrier signal is generated as O2 (S2902).

Subsequently, the feedback output FO2 (S2906) is added as a modulation signal to the above-mentioned O2 which is a carrier signal, and a new address is obtained and is made O2 (S290).
3). The feedback output FO2 is obtained by executing the process of S2906 described later at the previous interrupt timing.

Then, the value of the triangular wave corresponding to the addition address O2 is calculated. Actually, the external memory 11
6 (FIG. 1) stores the above-described triangular wave data,
The triangular wave data is obtained by table lookup at the address O2 (S2904).

Subsequently, the triangular wave data is multiplied by an envelope value E2 to obtain an output O2 (S2905).

Thereafter, a feedback signal is output to this output O2.
The level FL2 is multiplied to obtain a feedback output FO2 (S2907). In the case of the present embodiment, the output FO2 is used by the operator 2 (OP2) at the next interrupt timing.
Input to

The modulation level M is set to O2.
L2 is multiplied to obtain a modulation output MO2 (S2907). This modulation output M02 becomes a modulation input to the operator 1 (OP1).

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

First, the pitch data P1 is added to the current address A1 of the operator 1 (S2908), and the obtained value is subjected to the above-mentioned modified sine transform to change the carrier signal to O1.
(S2909).

Next, the above-mentioned modulation output MO2 is added to this O1 to form a new O1 (S2910). This value O1 is converted into a triangular wave (S2911), and further multiplied by an envelope value E1 to obtain a musical sound waveform output O1. Is obtained (S
2912).

This is the RAM 206 (FIG. 2) or 306
It is added to the buffer B (see FIG. 21) in FIG.
2913) The TM modulation processing for one sounding channel is ended.
The sound source processing according to the four methods of PCM, DPCM, FM, and TM has been described above. Among them, the two systems, FM and TM, are modulation systems.
Although the processing is performed by two operators based on the algorithm shown in FIG. 30, the sound source processing actually performed at the time of performance involves 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.

The FM system and T by the four operators in FIG.
FIGS. 32 and 3 show examples of normally considered sound source processing of the M system.
3 and FIG. 34 and FIG. <Second sound source processing by FM modulation method>

FIGS. 32 and 33 are operation flowcharts of sound source processing that can be normally considered based on the FM modulation method corresponding to the algorithm 1 of FIG. 31 (a). Each variable in the flow is stored in the RAM 206 of the MCPU 101 or the SCPU 102,
306 is stored in any of the sound channel areas (FIG. 17). Although the data does not correspond to each data of the FM format in the table 1 of FIG. 20, it is obvious that the data format can be easily realized by extending the data format.

First, the pitch data P4 is added to the current address A4 of the operator 4 (OP4) (S3201). Next, the feedback output FO4 (S32
05) is added as a modulation input, and a new address AM4 is obtained (S3202). Further, the value of the sine wave corresponding to the address AM4 (phase) is read from the external memory 116 (see FIG. 1) (S3203), and is multiplied by the envelope value E4 to obtain the output O4 (S3204). Thereafter, the output O4 is multiplied by the feedback level FL4 to obtain a feedback output FO4 (S3205). The output O4 is multiplied by the modulation level ML4 to obtain a modulation output MO4 (S3206). This modulation output MO4 becomes a modulation input to the next operator 3 (OP3).

Next, the processing shifts to the processing of the operator 3 (OP3).
This processing is almost the same as that of the above-described operator 4 except that there is no modulation input by the feedback output. First, the pitch data P3 is added to the current address A3 of the operator 3 (OP3) (S3207). Next, address A3
Is added as a modulation input to obtain a new address AM3 (S3208). Further, the value of the sine wave corresponding to the address AM3 (phase) is read from the external memory 116 (FIG. 1) (S3209), and is multiplied by the envelope value E3 to obtain the output O3 (S3210). Thereafter, the output O3 is multiplied by the modulation level ML3 to obtain a modulation output O3 (S3211). This modulation output MO3 is
Modulation input to the next operator 2 (OP2).

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

Finally, the process of the operator 1 (OP1) is started. In this case, the process up to step S3220 is the same as before. Then, the tone waveform output O1 obtained in S3220 is accumulated in the buffer B (see FIG. 21) as a carrier (S3221). <Second sound source processing by TM modulation method>

FIGS. 34 and 35 are operation flowcharts of sound source processing that can be normally considered based on the TM modulation method corresponding to the algorithm 1 in FIG. The variables in the flow are stored in the RAMs 206 and 3 of the MCPU 101 or the SCPU 102.
06 is stored in any of the sound channel areas of FIG. Although it does not correspond to each data of the TM format of the table 1 of FIG. 20, it is obvious that it can be easily realized by extending the data format of FIG.

First, the pitch data P4 is added to the current address A4 of the operator 4 (OP4) (S3401). Next, by the modified signature conversion fc, the above address A4
The modified sine wave corresponding to (phase) is stored in the external memory 116.
(FIG. 1) and a carrier signal is generated as O4 (S3402). Then, the feedback output FO4 (see S3407) is added to the output O4 as a modulation signal, and a new address O4 is obtained (S3403). Next, the value of the triangular wave corresponding to the address O4 (phase) is read from the external memory 116 (FIG. 1) (S3404), and is multiplied by the envelope value E4 to obtain the output O4 (S34).
05). Thereafter, the output O4 is multiplied by the modulation level ML4 to obtain a modulation output MO4 (S3406), and the output O4 is multiplied by the feedback level FL4 to obtain a feedback output FO4 (S3407). The modulation output MO4 serves as a modulation input to the next operator 3 (OP3).

Next, the processing of the operator 3 (OP3) is started.
This processing is almost the same as that of the above-described operator 4 except that there is no modulation input by the feedback output. First, the pitch data P is stored in the current address A3 of the operator 3.
3 is added (S3408), and the resulting value is subjected to a modified sine transform to obtain a carrier signal as O3 (S34).
09). Next, the above-mentioned modulation output M is applied to O3.
O4 is added to obtain a new O3 (S3410), and this value O
3 is converted into a triangular wave (S3411), and the envelope value E is further converted.
The output O3 is obtained by multiplying by 3 (S3412). Furthermore,
This is multiplied by the modulation level ML3 to obtain a modulation output MO3 (S3413). This modulation output MO3 becomes a modulation input to the next operator 2 (OP2).

Next, the processing of the operator 2 (OP2) is executed. This processing is the same as that of the above-described operator 3 except that the modulation input is different, so that 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 S3424. The tone waveform output O1 obtained in S3424 is accumulated as a carrier in the buffer B (see FIG. 21) (S3425).

Although the embodiment of the normal sound source processing by the modulation method has been described above, this processing is processing for one sounding channel as described above, and is actually the processing of the MCPU 101 and the SCPU 1.
For each CPU 02, eight sounding channels are processed (see FIG. 15). If the modulation method is specified for a certain sound channel, the sound source processing by the above-described modulation method operates. <First Modification Example of Modulation System>

Next, a first improved 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, 4
The processing has the same program structure except for the variable names used. Here, the operator processing cannot be performed 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, it is necessary to determine which output of the operator process is to be used as the modulation input, or whether to output the own modulation process by feeding back the output of the own operator process instead of the other operator processes. Therefore, in the operation flow of FIG. 36, such a connection relationship is collectively performed as an algorithm process (S3605), and the connection relationship obtained by this process is used for the modulation input in each operator process (S3602 to S3604) at the next interrupt timing. Act to determine. 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 processing and the algorithm processing are separated, the program of the operator processing may be the same for any algorithm, and only the algorithm processing 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 system based on the above basic concept will be described. FIG. 37 shows an operation flowchart of the operator process in FIG. 36 according to the FM modulation method, taking the operator 1 process as an example, and FIG. 38 shows an arithmetic algorithm per operator. The other operators 2 to 4 are completely the same except that the subscript numbers of the variables are different. Each variable in the flow is the MCPU 101 or SCPU
It is stored in any of the sound channels in FIG. 17 on the RAMs 206 and 306 of the CPU 102.

In FIG. 37, first, pitch data P1 is added to the address A1 corresponding to the phase angle, and a new address A1 is set (S3701).

Next, the modulation input MI1 is added to the address A1, and the address AM1 is obtained (S3702). The modulation input MI1 is determined by the algorithm processing S3605 (FIG. 36) at the previous interrupt timing, and depending on the algorithm, is the feedback output FO1 of the operator itself, or the output MO2 of another operator, for example, the operator 2 I do.

Next, the value of the sine wave corresponding to the address (phase) AM1 is read from the external memory 116 (FIG. 1), and the output O1 is obtained (S3703). After that, the value obtained by multiplying this by the envelope data E1 is given to the operator 1
(S3704).

Further, the output O1 is multiplied by the feedback level FL1 to obtain a feedback output FO1 (S3705), and the modulation level ML1 is applied to the output O1.
Is multiplied to obtain a modulation output MO1 (S3706).

Next, an improved example of the TM modulation system based on the above basic concept will be described. FIG. 39 shows an operation flowchart of the operator process of FIG. 36 according to the TM modulation method, taking the operator 1 process as an example, and FIG. 40 shows an arithmetic algorithm per operator. The other operators 2 to 4 are completely the same except that the subscript numbers of the variables are different. Each variable in the flow is the MCPU 101 or SCPU
It is stored in any of the sound channels in FIG. 17 on the RAMs 206 and 306 of the CPU 102.

In FIG. 39, first, the current address A1
Is added to the pitch data P1 (S3901).

Next, the modified sine wave corresponding to the address A1 (phase) is read from the external memory 116 (FIG. 1) by the modified sine transform fc, and the carrier signal is generated as O1 (S3902).

Then, the modulation input MI1 is added as a modulation signal to the output O1 to obtain a new address O1 (S1).
3903).

Next, the value of the triangular wave corresponding to the address O1 (phase) is read from the external memory 116 (S
3904), which is multiplied by the envelope value E1 and the output O1
(S3905).

Thereafter, a feedback signal is output to this output O1.
The level FL1 is multiplied to obtain a feedback output FO1 (S3906), and the output O1 is multiplied by the modulation level ML1 to obtain a modulation output MO1 (S3907).

FIG. 41 shows a specific example of the algorithm processing S3605 of FIG. 36 for determining the modulation input in the operator processing in both the FM system and the TM system.
And the operation flowchart of FIG. The flow of FIG. 13 is common to the FM system and the TM system, 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 (S4100).

First, Algorithm 1 is a 4-operator (abbreviated as OP) serial type as shown in FIG. 31 (a), and only OP4 has a feedback input. That is,

The feedback output FO4 of OP4 is used as the modulation input MI4 of OP4 (S4101).

The modulation output MO4 of OP4 is used as the modulation input MI3 of OP3 (S4102).

The modulation output MO3 of OP3 is used as the modulation input MI2 of OP2 (S4103).

The modulation output MO2 of OP2 is used as the modulation input MI1 of OP1 (S4104).

The algorithm is that the output O1 of OP1 is added to the buffer B (see FIG. 21) as a carrier output (S4105).

The algorithm 2 is as shown in FIG.
OP2 and OP4 have feedback inputs. That is,

The feedback output FO4 of OP4 is used as the modulation input MI4 of OP4 (S4106).

The modulation output MO4 of OP4 is used as the modulation input MI3 of OP3 (S4107),

The feedback output FO2 of OP2 is used as the modulation input MI2 of OP2 (S4108).

The modulation outputs MO2, OP2 of OP2 and OP3
MO3 is set as the modulation input MI1 of OP1 (S4109),

The algorithm is that the output O1 of OP1 is added to the buffer B as a carrier output (S4110).

In Algorithm 3, OP2 and OP4 have feedback inputs, and two 2-operator serial types are configured in parallel. That is,

The feedback output FO4 of OP4 is used as the modulation input MI4 of OP4 (S4111).

The modulation output MO4 of OP4 is used as the modulation input MI3 of OP3 (S4112).

The feedback output FO2 of OP2 is used as the modulation input MI2 of OP2 (S4113).

The modulation output MO2 of OP2 is used as the modulation input MI1 of OP1 (S4114).

The algorithm is that the outputs O1 and O3 of OP1 and OP3 are added to the buffer B as a carrier output (S4115).

Algorithm 4 is a 4-operator parallel type, in which all operators have feedback inputs. That is,

The feedback output FO4 of OP4 is used as the modulation input MI4 of OP4 (S4116).

The feedback output FO3 of OP3 is used as the modulation input MI3 of OP3 (S4117),

The feedback output FO2 of OP2 is used as the modulation input MI2 of OP2 (S4118).

The feedback output FO1 of OP1 is used as the modulation input MI1 of OP1 (S4119).

The algorithm is such that the outputs O1, O2, O3 and O4 of all the operators are added to the buffer B (S4120).

The sound source processing for one channel is completed by the above-described operator processing and algorithm processing, and if there is no change in the algorithm, sound generation (sound source processing) continues in this state. <Second Modified Modulation Example>

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

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

Therefore, in a second improved example described below, in FIG.
By further developing the first improved example of No. 6, only the operator processing is performed at a certain interrupt timing, and only the algorithm processing is performed at the next interrupt timing, so that the operator processing and the algorithm processing are alternately performed. As a result, the processing load per one interrupt timing can be significantly reduced. as a result,
One sample data is output for every two interrupts.

This operation will be described with reference to the operation flowchart of FIG. First, since the operator process and the algorithm process are performed alternately, Smod
It is determined whether or not 2 (“x mod y” is an operation indicating a remainder obtained by dividing the value x by the value y), that is, whether S is an even number (S4301). This variable S is M
Only one is provided on the RAM 206 of the CPU 101 (FIG. 21).

At a certain interrupt timing, S
When mod 2 is 0, the process enters the operator process route, and the processes of the operators 1 to 4 are executed (S4302 to S4305).
This processing is the same as in FIG. 37, FIG. 38 or FIG. 39, FIG.

Next, through the route of the operator process, an output process is executed in which the value of the buffer BF (in the case of the FM system) or the buffer BT (in the case of the TM system) is set in the buffer B (S4310). The buffer BF or BT is provided for each sounding channel, and the MCPU 101 or the SCPU 1
This is stored for each sound channel area in FIG. 17 on the RAMs 206 and 306 (see FIG. 21). The buffer BF or BT stores the waveform output value after the algorithm processing (to be described later with reference to FIGS. 44 and 45), but at the current interrupt timing, the algorithm processing is not executed and the contents of the buffer BF or BT are stored. Is not changed, the same waveform output value as the previous interrupt timing 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 in the current processing of the operators 1 to 4 is transferred to the MCPU until the next interrupt timing.
It is stored in each sound channel area of FIG.

When the next interrupt occurs, first, the content of the variable S is incremented by 1 at the beginning of the MCPU interrupt processing of FIG. 10 (S1010). Therefore, as described above, if Smod 2 is 0 at the previous interrupt timing, Smod 2 becomes 1 at the current interrupt timing, and the algorithm process is executed (S4306).

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. 17 is used, and the modulation input for the next operator processing is performed. A decision process 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. Specific examples of the algorithm processing are shown in the operation flowcharts of FIGS. 44 and 45. In this flow, the same processes as those in FIGS. 41 and 42 are performed in steps denoted by the same reference numerals as those in FIGS. 41 and 42. What differs from the cases of FIGS. 41 and 42 is the output portion of S4401 to S4404. Here, in the case of the algorithm 1 and the algorithm 2, the contents of the output O1 of the processing of the operator 1 are directly stored in the buffer BF.
Alternatively, it is held in the BT (S4401, S4402). In the case of the algorithm 3, the value obtained by adding the output O3 to the output O1 is held in the buffer BF or BT (S4403).
Further, in the case of Algorithm 4, the outputs O2 and O
3, the value obtained by adding O4 is held in BF or BT (S
4404).

As described above, based on the processing in step S1010 in FIG.
Becomes even or odd, and the operator process and the algorithm process are alternately executed at every other interrupt timing based on the determination process of step S4301 in FIG. 43.
The processing load of the sound source processing program per one interrupt timing can be significantly reduced. In this case, since it is not necessary to lengthen the interrupt cycle, the processing load can be reduced without affecting the program operation without increasing the interruption time of the main operation flowchart in FIG. 9 due to the interrupt. Therefore, for example, the interval of taking in a keyboard key executed according to FIG. 9 does not become long, and the response performance as an electronic musical instrument is not affected.

Next, in the case where the second improved example of the modulation system shown in FIGS. 43 to 45 is used as one of a plurality of sound source systems to be executed as sound source processing, the main flow chart of FIG. The process of setting the pitch data to the tone generation channel, which is executed as the process of step S909 in step S909, will be described.

In the main flowchart of FIG.
As described above, when the key assignment process based on the algorithm of FIG. 13 is executed in steps S905 to S907, normally in subsequent step S909, pitch data is set to the assigned sounding channel, and furthermore, The variable H corresponding to the tone generation channel for which the pitch data has been set in the flow one round preparation process of step S910.
One of SS1 to HSS8 is set to 1. As a result, based on the algorithm of FIG.
In the CPU interrupt processing or the SCPU interrupt processing of FIG. 11, the sound source processing of step S1011 or S1101 is executed, and a sound generation operation is performed.

Here, when the sound source processing is processing based on the second improved example of the modulation method shown in FIGS. 43 to 45, as described above, the operator processing and the algorithm processing are performed for each interrupt timing. Executed alternately.

Therefore, if the pitch data is set for a certain sounding channel and the algorithm processing is executed on the sounding channel at the interrupt timing immediately after that, the operator processing is still performed once for the sounding channel after the sounding is started. Is not executed, the algorithm processing is executed, and the processing becomes useless.

Therefore, in the above-described embodiment according to the second modification of the modulation method, a new assignment is generated for a certain sound channel, and the sound source processing to be executed in the sound channel is the modulation method. If the immediately following interrupt timing is the timing of the algorithm processing, the pitch data is not set to the sounding channel, and control is performed so as to wait until the next timing when the operator processing is executed. As a result, the variable HSS is not set in step S910 in FIG. 9 until the next time the operator process is executed for the sounding channel, so that even if an interrupt occurs, the algorithm in FIG. The sound source processing can be controlled so as not to be executed, and the execution time of the interrupt processing can be reduced.

FIG. 46 shows FIG. 9 for realizing the above functions.
9 shows an operation flowchart of pitch data setting processing executed in step S909 of FIG.

First, it is determined whether or not an assignment has been made to a new tone generation channel by the key assignment processing based on the algorithm of FIG. 13 in steps S905 to S907 of FIG. 9 (S4601).

If this determination is YES, it is further determined whether or not the sound source processing to be executed in the sound channel is a modulation (FM or TM) method (S4602). This determination is based on the tone generator method No. of the data loaded in the channel area to be assigned in FIG. 17 (FIGS. 19 and 2).
0).

If the determination is YES, it is determined whether Smod2 of the variable S is 1 (S4603). If the determination is NO and the value of the variable S is currently an even number, an interrupt is generated and step S in FIG.
In 1010, the variable S is set to an odd number, and the determination in S4603 is YES.
(Until S4603 is repeated).

Then, when the variable S becomes an odd number, step S
At the timing when the determination of 4603 becomes YES, the RAM 20
The pitch data based on the key-on note number or the like is set in an area (P1, P2 in FIGS. 17 and 20) corresponding to the newly assigned sounding channel in No. 6 or 306 (S4604).

When the pitch data is set in step S909 in FIG. 9 as described above, in step S910 in FIG. 9, the variable HSS corresponding to the sound channel is set.

As a result, if an interrupt occurs immediately after that, the variable S is set to an even number in step S1010 in FIG. 10, and the algorithm in FIG. 15 is executed as the sound source processing in step S1011 in FIG. 10 or step S1101 in FIG. For example, if it is the first sounding channel, FIG.
The determination in step S1502 of step 5 is YES, and the
The operation flowchart of the second modified example of the modulation method shown in FIG. 43 is executed as the sound source processing of 1503, and the operator processing is always executed in step S4301.

Conversely, even if an interrupt occurs immediately after the determination in S4603 is NO, since the pitch data has not been set for the sounding channel, the determination in step S1502 and the like in FIG. 15 is NO. The sound source processing for the sound channel is skipped, and the execution time of the interrupt processing can be reduced.

The operation of generating tone data for each sound channel by sound source processing of software based on various sound source systems has been described above. <Function key processing>

Next, the specific operation of the function key processing (S903) in the main operation flowchart of FIG. 9 when playing an actual electronic musical instrument will be described.

In the sound source processing performed for each sounding channel described above, the MCPU 101 is connected to, for example, an operation panel of an electronic musical instrument via the input port 210 (see FIG. 2) of the MCPU 101 by a function key 701 shown in FIG. Or SCP
Each sound channel area (see FIG. 17) on the RAM 206 or 306 (see FIGS. 2 and 3) of the U 102 has a data format (see FIGS.
Reference) is set.

FIG. 47 is a diagram showing an example of the arrangement of a part of the function keys 701 in FIG. In the figure, a part of the function key 701 is realized as a tone designation switch, and when a switch such as “piano”, “guitar”,. Is selected,
The guide lamp lights up. The DPCM / TM system selection switch 4701 changes the tone of these instrument sounds to DP
The sound source method of the CM method or the TM method is selected.

On the other hand, when the switch of the “Tuba” in the B group is pressed, the FM system is used. When the “Base” is pressed, the PCM / TM system is used.
Is pressed, the respective timbres are designated by the PCM method, and musical tones based on those sound source methods are generated.

FIGS. 48 and 49 show the RAM 206 when the "piano" and "bass" switches are pressed.
Alternatively, an example of assigning a sound source system to each sounding channel area shown in FIG. In the case of “piano”, as shown in FIG. 48, all of the eight tone polyphonic sounding channels of the MCPU 101 and the SCPU 102 are assigned DP.
In the case where the CM system is assigned, and in the case of “base”, as shown in FIG.
The TM system is assigned to each even-numbered sounding channel. With this, 2 based on PCM system and TM system
A musical sound waveform for one sound is obtained as a mixture of musical sound waveforms generated in the sounding channel. In this case, each C
Four-tone polyphonic per PU, eight-tone polyphonic in total for two CPUs.

FIGS. 50 and 51 are a part of the operation flowchart of the function key processing of S903 in the main operation flowchart of FIG. 9, and are the operation flowcharts of the processing for the tone color designation switch group of FIG.

First, it is determined whether or not the player has operated the DPCM / TM switch 4701 (S5001).
If the determination is YES, it is determined whether the variable M is zero (S5002). The variable M is the RAM 206 of the MCPU 101
(FIG. 2) One common channel is reserved for all sound channels (FIG. 2).
1), the value 0 in the case of the DPCM system, and the value 1 in the case of the TM system.

In step S5002, the value of the variable M is 0 and the determination is YE.
In the case of S, the value 1 is set to the variable M (S5003).
This is the state where the DPCM method has been selected.
This means that the / TM switch 4701 has been pressed and the TM system has been selected. If the value of the variable M is 1 in S5002 and the determination is NO, the value 0 is set to the variable M (S5004). This means that the DPCM / TM switch 4701 has been pressed and the DPCM system has been selected in the state where the TM system has been selected.

Next, it is determined whether or not the tone color of the group A in FIG. 47 is currently designated (S5005). DPC
Since the M / TM switch 4701 is effective only for the tone color of the group A, the tone color of the group A
Only when the determination of 5005 is YES, D of S5006 to S5008
An operation corresponding to PCM / TM switch 4701 is executed.

In S5006, it is determined whether or not the variable M is 0.

If the determination in S5006 is YES, the DPCM
Since the DPCM method is selected by the / TM switch 4701, the RAMs 206 and 306 (see FIGS. 2 and 3)
Data is set in each of the sounding channel regions in the DPCM format shown in FIG. That is, the DP is added to the head area G (see the column of DPCM in FIG. 19) of each sounding channel area.
A sound source method No. indicating the CM method is set. Subsequently, various parameters corresponding to the currently designated timbre are set in the second and subsequent areas of each sounding channel area (S5007).

If the determination in S5006 is NO, DP
Since the TM system is selected by the CM / TM switch 4701, data is set in each sounding channel area in the TM format shown in FIG. That is, first, a sound source system number indicating the TM system is set in the head region G 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 (S5008).

The above is the case where the DPCM / TM switch 4701 in FIG. 47 has been operated. However, when the switch is not operated and the judgment in step S5001 is NO, or when the tone color of the A group is designated. If the determination in step S5005 is NO, the processing from switch S5009 is performed.

First, in step S5009, it is determined whether or not the tone color switch shown in FIG. 47 has changed (S5009).

If the determination is NO, there is no need to perform the processing for the timbre switch, and the function key processing (S903 in FIG. 9) is terminated.

If there is a change in the tone color switch and the determination in S5009 is YES, it is next determined whether or not the tone color of Group B has been designated (S5010).

If the timbre of group B is designated and the determination in S5010 is YES, the tone generator data format corresponding to the designated timbre is stored in each sounding channel area on RAMs 206 and 306 (see FIGS. 2 and 3). Is used to set the data. Then, the leading area G of each sounding channel area
(Refer to FIG. 19 and FIG. 20).
Is set. Also, various parameters corresponding to the currently designated tone color are set in the second and subsequent areas of each sounding channel area (S5011). For example, when the base switch of FIG. 47 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. Each is set.

If the tone switch of the group A is designated and the judgment in S5010 is NO, it is judged whether or not the variable M is 1 (S5012). If the TM method is currently selected and the determination in S5012 is YES, data is set in each sounding channel area in the TM format of FIG. 20 in the same manner as in step S5008 (S5013).

If the DPCM method is selected and S50
If the determination in step 12 is NO, the DPC of FIG. 19 is added to each sounding channel area in the same manner as in step S5007.
Data is set in the M format (S5014). <First Embodiment of Keyboard Key Processing at Key Depression>

Next, the specific operation of the keyboard key processing (S905) in the main operation flowchart of FIG. 9 when playing an actual electronic musical instrument will be described.

First, a first embodiment of the 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 color of the group A in FIG. 47 is designated, it is connected via the input port 210 (see FIG. 2) of the MCPU 101 in FIG. Alternatively, the R position of the MCPU 101 or the SCPU 102 depends on the keyboard position when the keyboard key 702 shown in FIG.
The sound source system set in each sound channel area (see FIG. 17) on the AM 206 or 306 (see FIGS. 2 and 3) is automatically switched. In this case, with the key code numbers 31 and 32 of the keyboard key 701 shown in FIG.
The DPCM method is assigned when the key code of the depressed key is the low tone range of 31 or less, and the TM method is assigned when the key code is the high tone range of 32 or more. Note that FIG.
No special keyboard key processing is performed when the tone color of the B group is designated.

FIG. 52 is a part of the operation flowchart of the keyboard key processing of S905 in the main operation flowchart of FIG. 9, and is a flow of the first embodiment of the processing at the time of depressing the keyboard key 701 of FIG. .

First, it is determined whether or not the tone color of the group A in FIG. 47 is currently specified (S5201).

If this determination is NO and the tone color of group B is currently specified, the special processing in FIG. 52 is not performed.

If the determination in S5201 is YES and the tone color of Group A is currently designated, the key of the key determined to be "key pressed" in the keyboard key loading process of S904 in the main operation flowchart of FIG. Code is 31
It is determined whether the number is less than or equal to the number (S5202).

If the low tone range 31 or lower is depressed and the determination in S5202 is YES, it is determined whether or not the variable M is the value 1 (S5203). The variable M is set in the operation flowcharts of FIGS. 50 and 51 which are a part of the function key processing of S903 in the main operation flowchart of FIG. 9. As described above, the value of the variable M is It takes a value of 0 and a value of 1 in the case of the TM system.

If the determination in S5203 is YES (M = 1) and the TM system tone generator system is currently specified, the RAM is changed to the DPCM system, which is the lower tone range tone generator system.
In the sound channel area to which the depressed key on 206 or 306 (see FIGS. 2 and 3) is assigned, the DP of FIG.
Data is set in the CM format. That is, the sound source system No. indicating the DPCM system is set in the head area G (see the column of DPCM in FIG. 19) 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 (S5204). Then, the value 1 is set to the flag C.
Is set (S5205). The flag C indicates that the MCPU 101
This flag C is used in a key release process described later with reference to FIG. 54, although it is a variable secured in each sounding channel area on the RAM 206 (see FIG. 2).

On the other hand, in step S5202, if the treble range of No. 31 or higher is depressed and the determination is NO, the variable M
Is determined to be the value 1 (S5206). S5206
Is NO (M = 0) and if the sound source system of the DPCM system is currently specified, the pressed key on the RAM 206 or 306 is changed in order to change to the TM system which is the sound source system of the high frequency range. Is assigned to the sound channel area to which
Data is set in a TM format of 0. That is, the tone generator method No. indicating the TM method is set in the head area G (see the column of TM in FIG. 20) 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 (S5207). Thereafter, the value 2 is set in the flag C (S5205).

In the above processing, if the determination in step S5203 is NO and if the determination in step S5206 is YES, the desired sound source method is used, and no special processing is performed. <Second embodiment of keyboard key processing at 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 color of the group A in FIG. 47 is designated, the keyboard is connected via the input port 210 of the MCPU 101 (see FIG. 2). Key 70 shown in FIG. 7 or FIG.
The tone generation method set in each sound channel area (see FIG. 17) on the RAM 206 or 306 (see FIGS. 2 and 3) of the MCPU 101 or the SCPU 102 depends on the key pressing speed, that is, the velocity when the key 2 is pressed. It is switched automatically. In this case, the velocity value is MIDI
(Musical Instrument Digital Interface) If the velocity value of the key depressed is 64 or more at the time of a key depressing operation at 64, which is 1/2 of 127 of the maximum value of the standard, DP
The CM method is assigned to the CM method, and the TM method is assigned to a slow key press operation with a velocity value of 64 or less. FIG.
When the tone color of the B group 7 is designated, no special keyboard key processing is executed. FIG. 53 is a part of an operation flowchart of the keyboard key processing of S905 in the main operation flowchart of FIG.
9 is a flowchart of a second example of a process at the time of key depression of No. 1;

First, it is determined whether or not the tone color of the group A in FIG. 47 is currently specified (S5301).

If this determination is NO and the tone color of group B is currently specified, the special processing shown in FIG. 53 is not performed.

If the determination in S5301 is YES and the tone color of Group A is currently specified, the velocity of the key determined to be "key pressed" in the keyboard key loading process of S904 in the main operation flowchart of FIG. But 64
It is determined whether this is the case (S5302). In addition, 64 of this velocity is mp (mezzo piano) of MIDI standard
Is equivalent to

If the velocity value is 64 or more and the determination in S5302 is YES, it is determined whether or not the variable M is the value 1 (S5303). The variable M is a part of the function key processing of S903 in the main operation flowchart of FIG.
51, and is set in the operation flowchart of FIG. 51. As described above, the value of the variable M takes the value 0 in the DPCM system and the value 1 in the TM system.

If the determination in S5303 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 for a fast key press operation, FIG. As in the process of S5204 in the case of the first embodiment, data in the DPCM format of FIG. 19 is stored in the sound channel area on the RAM 206 or 306 (see FIGS. 2 and 3) to which the depressed key is assigned. Is set (S53
04), the value 1 is set to the flag C (S5305).

On the other hand, in step S5302, if the velocity value is smaller than 64 and the determination is NO, the variable M
Is determined to be 1 (S5306).

[0271] If the determination in S5306 is NO (M = 0),
When the sound source system of the DPCM system is specified, the sound source system is changed to the TM system which is the sound source system at the time of the slow key depression operation.
As in the case of the process of S5207 in the first embodiment of the second example, data is set in the TM format of FIG. 20 in the sounding channel area on the RAM 206 or 306 to which the depressed key is assigned ( (S5307), the value 2 is set to the flag C (S5308).

In the above processing, if the determination in step S5303 is NO and if the determination in step S5306 is YES, the desired sound source method is used, and no special processing is performed. <Example of keyboard key processing at key release>

Next, an embodiment of keyboard key processing at the time of key release will be described.

According to the first or second embodiment of the keyboard key processing at the time of key depression, the tone generator system can be automatically changed according to the key range (tone range) and velocity. It needs to be restored. This is realized by the embodiment of the keyboard key processing at the time of key release described below.

FIG. 54 is a part of the operation flow chart of the keyboard key processing of S905 in the main operation flow chart of FIG. 9, and is a flow of processing when the key 701 of FIG. 7 is released.

First, the RAM 206 or 306 to which the key determined to be "key released" in the keyboard key loading process of S904 in the main operation flowchart of FIG. 9 is assigned.
The value of the flag C set in the sounding channel area (see FIGS. 2 and 3) is determined. Now, the flag C is set to S5205 and S5208 in FIG. 52 or S5305 or S5308 in FIG.
The initial value at the time of key depression is 0,
When the sound source system is changed from the TM system to the DPCM system when the key is pressed, a value of 1 is set, and the DPCM system is changed to the TM system.
When the sound source system is changed to the system, the value 2 is set. Therefore, if no change is made at the time of key depression, the initial value remains at 0.

Then, in the key release process shown in FIG.
If the value of the flag C is 0 in the determination in step S5401, the sound source method has not been changed by the key range or velocity, so that normal key release processing is performed without performing any special processing.

If it is determined in step S5401 that the value of the flag C is 1, the sound source system is changed from the TM system to the DP system when the key is pressed.
It has been changed to the CM system. Therefore, in order to return to the TM system, data is set in the TM format of FIG. 20 in the tone generation channel area on the RAM 206 or 306 (see FIG. 2 or 3) to which the depressed key is assigned.
That is, the tone generator method No. indicating the TM method is set in the head area G (see the column of TM in FIG. 20) 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 (S5402).

On the other hand, if it is determined in step S5401 that the value of the flag C is 2, the sound source method is set to DPCM when the key is pressed.
The system has been changed from the TM system to the TM system. So, DPCM
In order to return to the system, the sounding channel area to which the depressed key is assigned on the RAM 206 or 306 is placed in FIG.
Is set in the DPCM format. That is, the head area G of each sounding channel area (DPC in FIG. 19)
The sound source method No. indicating the DPCM method is set in the M column). Subsequently, various parameters corresponding to the currently designated timbre are set in the second and subsequent areas of each sounding channel area (S5403).

After the above operation, the value of the flag C is returned to 0, and the processing in FIG. 54 is terminated. Then, a normal key release processing (not shown) is executed. <An aspect of another embodiment>

In the series of embodiments of the present invention described above, as shown in FIG.
Although the two CPUs 102 process different sounding channels, the number of CPUs may be one or three or more.

The control ROM 201 shown in FIGS.
And 301 and the external memory 116 are constituted by a ROM card or the like, it is possible to provide various sound source systems to the user by the ROM card.

Further, the input port 2 of the MCPU 101 in FIG.
Various operating units can be connected to the unit 10 in addition to the musical instrument operating units as shown in FIGS. 7 and 8, thereby realizing various types 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 possible in which the tone generator is assigned to each sounding channel by the function key 701 in FIG. 7 or the keyboard key 702 in FIG.

As the modulation system, various systems can be applied in addition to the FM and TM systems.

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

[0287]

【The invention's effect】

According to the present invention, when sound source processing is executed using a plurality of processors, the load on each processor can be equalized, and the time required for the entire sound source processing per timing can be shortened. Becomes possible.

Further, when the sound source processing is realized by an interrupt program for the performance information processing program, the interrupt time is shortened, so that a lot of time can be allocated to the performance information processing program, and sophisticated and complicated performance information processing, etc. Can be executed in real time.

[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.

FIG. 4 is a configuration diagram of a conventional D / A converter unit.

FIG. 5 is a configuration diagram of a D / A converter unit according to the present embodiment.

FIG. 6 is a timing chart in D / A conversion.

FIG. 7 is a layout diagram of function keys and keyboard keys.

FIG. 8 is an explanatory diagram of a keyboard key.

FIG. 9 is a flowchart of a main operation.

FIG. 10 is an operation flowchart of an MCPU interrupt process.

FIG. 11 is an operation flowchart of an SCPU interrupt process.

FIG. 12 is a conceptual diagram showing a relationship between a main operation flowchart and an interrupt process.

FIG. 13 is an operation flowchart (part 1) of a key assignment process.

FIG. 14 is an operation flowchart (part 2) of a key assignment process.

FIG. 15 is an operation flowchart of a sound source process.

FIG. 16 is a diagram showing variables MEC and SEC on a RAM 206.

FIG. 17 is a diagram showing a storage area for each sounding channel on a RAM.

FIG. 18 is a conceptual diagram when a sound source processing method of each sounding channel is selected.

FIG. 19 is a configuration diagram (part 1) of a data format for each sound source method on a RAM.

FIG. 20 is a configuration diagram (part 2) of a data format for each sound source method on a RAM.

FIG. 21 is a diagram showing a buffer area on a RAM.

FIG. 22 is an operation flowchart of a sound source process according to the PCM method.

FIG. 23 is an operation flowchart (part 1) of sound source processing by the DPCM method.

FIG. 24 is an operation flowchart (part 2) of sound source processing by the DPCM method.

FIG. 25 is an explanatory view of the principle in the case where an interpolated value XQ is obtained by using the difference value D and the current address AF in the PCM method.

FIG. 26 shows a difference value D and a current address AF in the DPCM method.
FIG. 4 is a diagram for explaining the principle in a case where an interpolation value XQ is obtained by using the following equation.

FIG. 27 is an operation flowchart of a first sound source process by the FM method.

FIG. 28 is a diagram illustrating an algorithm of a first sound source process by the FM method.

FIG. 29 is an operation flowchart of a first sound source process according to the TM method.

FIG. 30 is a diagram showing an algorithm of the first sound source processing by the TM method.

FIG. 31 is a diagram illustrating an example of an algorithm in a modulation scheme.

FIG. 32 is an operation flowchart (part 1) of the second sound source processing by the FM method.

FIG. 33 is an operation flowchart (part 2) of the second sound source processing by the FM method.

FIG. 34 is an operation flowchart (part 1) of the second sound source processing by the TM method.

FIG. 35 is an operation flowchart (part 2) of the second sound source processing by the TM method.

FIG. 36 is an operation flowchart of a first modification of the modulation method.

FIG. 37 is an operation flowchart of an operator 1 process by the FM method according to the first improved example.

FIG. 38 is a diagram illustrating a calculation algorithm per operator in an operator 1 process by the FM method according to the first improved example.

FIG. 39 is an operation flowchart of an operator 1 process by the TM method according to the first improved example.

FIG. 40 is a diagram showing a calculation algorithm per operator in an operator 1 process by the TM method according to the first improved example.

FIG. 41 is an operation flowchart (part 1) of an algorithm process according to the first improved example;

FIG. 42 is an operation flowchart (part 2) of the algorithm processing according to the first improved example;

FIG. 43 is an operation flowchart of a second modification of the modulation method.

FIG. 44 is an operation flowchart (part 1) of an algorithm process according to the second improved example.

FIG. 45 is an operation flowchart (part 2) of the algorithm processing according to the second improved example;

FIG. 46 is an operation flowchart of pitch data setting processing according to a second modification of the modulation scheme.

FIG. 47 is a layout view of a part of function keys.

FIG. 48 is a diagram showing an example of assigning a sound source method to a sound channel in a PIANO (A group) of DPCM.

FIG. 49 is a diagram showing an example of assigning sound source systems to sounding channels in BASS (B group).

FIG. 50 is an operation flowchart (part 1) of a function key process.

FIG. 51 is an operation flowchart (part 2) of a function key process.

FIG. 52 is an operation flowchart of a first embodiment of keyboard key processing at the time of key depression.

FIG. 53 is an operation flowchart of a second embodiment of keyboard key processing at the time of key depression.

FIG. 54 is an operation flowchart of an embodiment of keyboard key processing at the time of key release.

[Explanation of symbols]

Reference Signs List 101 master CPU 102 slave CPU 103 MCPU external memory access address latch unit 104 SCPU external memory access address latch unit 105 access address conflict avoidance circuit 106 external memory selector unit 107 Left D / A converter unit 108 Right D / A conversion Unit 109 input port 110 output port 111, 112 output terminal 113 left output terminal 114 right 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 Multiplier 210 Input port 211 Output port 212 SCPU Internal RAM addressing bus interface 213 Write data bus interface to SCPU 214 Read data bus interface from SCPU 215 Address bus interface for external memory access 216 External memory data bus interface 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 interface to MCPU 312 Address bus interface for external memory access 313 External memory data bus -Interface 401, 501 Latch 402 D / A converter 701 Function key 702 Keyboard key ASC PU reset release signal (processing start signal) B SCPU processing end signal Ma SCPU Internal RAM addressing bus Dout Data bus that MCPU writes to SCPU Din Data bus that MCPU reads to SCPU MA MCPU Address bus where MCPU specifies external memory SA SCPU is Address bus that specifies external memory MD Data bus that MCPU reads from external memory Data bus that SD SCPU reads from external memory

Continuing on the front page (72) Inventor Kosuke Shinami 3-2-1, Sakaemachi, Hamura-cho, Nishitama-gun, Tokyo Casio Computer Co., Ltd. Hamura Technical Center (72) Inventor Koichiro Taiki 3-chome, Sakaemachi, Hamura-cho, Nishitama-gun, Tokyo No. 2 Casio Computer Co., Ltd. Hamura Technology Center (72) Kazuo Ogura 3-2-1-1 Sakaemachi, Hamura Town, Nishitama-gun, Tokyo Casio Computer Co., Ltd. Hamura Technology Center (58) Field surveyed (Int .Cl. ⁶ , DB name) G10H 7/02 G10H 1/02

Claims (2)

(57) [Claims] 1. A data storage means for storing tone control data for each of a plurality of sounding channels, and a sound source processing program for obtaining a tone signal at predetermined time intervals, and executing a sound source processing program for each sounding channel. A plurality of sound source processors each including a program execution means for generating a tone signal based on the corresponding tone control data, and a performance information processing program for processing performance information, and executing Controlling the tone control data in the data storage means; in this case, controlling the tone control data in each of the tone generation channels so that tone generation is performed by using each tone generation channel of each of the tone generator processors equally; Performance information for causing the program execution means of each sound source processor to execute the sound source processing program in parallel at time intervals; And an information processing processor. 2. The performance information processing processor is also used by one of the plurality of sound source processing processors. The one sound source processing processor executes the performance information processing program at normal times, and executes the performance information processing program at regular intervals. The interrupt processing causes the sound source processing processors to execute the sound source processing programs in parallel, and upon completion of the execution, returns control to the performance information processing program again to execute the program. 2. The musical sound waveform generator according to 1.