KR950006029B1 - Musical sound waveform generator - Google Patents

Abstract

No content.

Description

Acoustic Frequency Generator

1 is an overall configuration diagram of an embodiment according to the present invention.

2 is an internal configuration diagram of the master CPU.

3 is an internal configuration diagram of a slave CPU.

4 (a)-(b) are an overall operation flowchart of this embodiment.

5 is a conceptual diagram showing a relationship between a main operation flowchart and interrupt processing.

6 (a) is a block diagram of a conventional D / A converter unit.

6 (b) is a block diagram of a D / A converter section according to the present embodiment.

7 is a timing chart for D / A conversion.

8 (a) is a layout view of a function key and a keyboard key.

Fig. 8 (b) is an explanatory diagram of the keyboard keys.

9 is a diagram showing a storage region for each pronunciation channel on a RAM.

10 is a conceptual diagram when selecting a sound source processing method of each pronunciation channel.

11 is a configuration diagram of a data format for each sound source method on a RAM.

12 is a diagram showing a buffer area on a RAM.

13 is an operation flowchart of sound source processing by the PCM method.

14 is an operation flowchart of sound source processing by the DPCM method.

15 (a) and 15 (b) are explanatory diagrams for calculating the interpolation value XQ using the difference value D and the current address AF.

Fig. 16 (a) is an operational flowchart of the sound source processing (No. 1) by the FM system.

16 (b) is a diagram showing an algorithm of sound source processing (No. 1) by the FM system.

Fig. 17 (a) is an operational flowchart of sound source processing (No. 1) by the TM system.

Fig. 17B is a diagram showing an algorithm of sound source processing (No. 1) by the TM system.

18 shows an example of an algorithm in the modulation scheme.

FIG. 19 is an operational flowchart of the sound source processing (No. 2) by the FM system. FIG. 20 is an operational flowchart of the sound source processing (No. 2) by the TM method.

21 is an operation flowchart of the first improvement example of the modulation method.

Fig. 22A is an operational flowchart of operator 1 processing by the FM system according to the first improvement example.

FIG. 22 (b) is a diagram showing a calculation algorithm per operator of one operator processing by the FM system according to the first improvement example. FIG.

23 (a) is an operation flowchart of operator 1 processing by the TM method according to the first improvement example.

FIG. 23 (b) shows a calculation algorithm per operator in one operator processing by the TM method according to the first improvement example.

24 is an operation flowchart of an algorithm processing according to the first improvement example.

25 is an operation flowchart of a second improvement example of the modulation method.

26 is an operational flowchart of an algorithm processing according to the second improvement example.

27 is a layout view of a part of a function key;

28 (a) and 28 (b) show examples of allocation of a sound source system to a pronunciation channel;

29 is an operation flowchart of function key processing.

30 is an operational flowchart of the first embodiment of keyboard key processing at the time of pressing.

Fig. 31 is an operational flowchart of the second embodiment of keyboard key processing at the time of pressing.

32 is an operational flowchart of an embodiment of keyboard key processing in this case.

The present invention relates to a sound source processing method in a sound wave generation process.

With the development of digital signal processing technology and LSI processing technology, various high-performance electronic musical instruments are realized. In particular, the emergence of electronic musical instruments having a modulated acoustic waveform generator which obtains variously modulated waveforms and outputs them as acoustic waveforms by applying modulation to an input signal generated due to performance information. Contributes to a significant increase.

Since the acoustic wave generator of an electronic musical instrument has a large amount and high-speed digital operation, it is conventionally used in a dedicated sound source circuit that realizes an architecture equivalent to the sound generation algorithm due to the sound source method required by hardware. It is comprised by. By such a sound source circuit, a sound source method resulting from the modulation method is realized.

As described above, the sound source circuit has a large circuit size for any sound source system. In the case of LSI, the size of the microprocessor for general-purpose data processing is about twice as large as the memory portion for storing the waveform data for modulation and the like. This is because the sound source circuit requires complicated address control for accessing waveform data due to various performance information. This is because a register or the like for temporarily holding intermediate data obtained in the process of sound source generation needs to be arranged everywhere with an architecture corresponding to the sound source method. In particular, the modulation method can change various algorithms of sound generation, and a hardware configuration corresponding to them is also required.

This is because a shift register or the like for performing sound source processing in hardware in time division for realizing a polyphonic configuration capable of sounding in parallel with a plurality of musical sounds is also required everywhere.

As described above, since the conventional sound wave waveform generator is constituted by a dedicated sound source circuit corresponding to the sound source system, the hardware scale becomes large, and when the product is realized by LSI, it is a product for raw materials in manufacturing LSI chips. In terms of ratio and the like, there is a problem of causing a cost up in the manufacturing stage and causing an increase in the size of the acoustic wave generator.

In addition, in the case where the sound source system is to be changed, the polyphonic number is to be changed, etc., a significant change of the sound source circuit is inevitably made, resulting in a cost up in the development stage.

In addition, when a conventional acoustic wave generator is realized as an electronic musical instrument, a microprocessor for generating data which can be processed by a sound source circuit in performance information corresponding to a performance operation, or performing performance information communication with other musical instruments. There is a need for a control circuit constituted by, for example. In addition, in such a control circuit, in addition to the performance information processing program for processing the performance information, a sound source control program corresponding to the sound source circuit for supplying data corresponding to the performance information to the sound source circuit is required. It is necessary to run the program in synchronization. In the complexity of such a program, there is a problem that it causes a large cost up in the development.

On the other hand, in recent years, many high-performance microprocessors for performing general-purpose data processing have been realized, and thus it is conceivable to realize a sound wave waveform generating apparatus which softly performs sound source processing using the microprocessor. However, there is no known technique for synchronously operating a sound source processing program for executing performance information processing for processing performance information. In particular, since the processing time changes in the sound source processing program by the sound source method, a complicated timing control program for outputting the generated sound data to the D / A converter is required. As described above, simply performing the sound processing in a simple manner makes the processing program very complicated, and in terms of processing speed and program capacity, processing of a high sound source method such as a modulation method cannot be performed. In particular, as described above, the modulation scheme may allow various algorithms for sound generation, but it is disadvantageous in terms of program capacity if the microprocessor individually has a sound source processing program corresponding to each of these algorithms. This, in turn, leads to cost up and enlargement of the sound generating apparatus.

The present invention does not require a dedicated sound source circuit, and an object thereof is to enable high-level sound source processing by a modulation method by program control of a microprocessor.

It is another object of the present invention to enable high-level sound source processing by a modulation method that operates with various algorithms for sound generation by program control of a microprocessor without requiring a dedicated sound source circuit.

According to the first state of the present invention, there is provided a memory means for storing a sound source processing program according to a modulation method, and a sound sound having a processing method of a modulation method by executing the sound source processing program stored in the memory means. There is provided a sound wave waveform generating apparatus, comprising: a signal generating means and a sound signal output means for outputting the sound signal generated by the sound signal generating means at a constant output time interval.

According to the sound wave waveform generating apparatus according to the first state of the present invention, it is possible to realize the high-level sound source processing according to the modulation method without using a dedicated sound source circuit, and also, by the movement of the sound signal output means, Since the output rate is kept constant, it is possible to obtain a sound wave without distortion.

According to the second state of the present invention, there is provided a performance information processing program for processing performance information, program storage means for storing a sound source processing program by a modulation method for obtaining a sound signal, and an address for controlling the program storage means. Controlling address control means, data storage means for storing necessary sound generation data for generating a sound signal in said modulation method, arithmetic processing means, said address control means, said data storage means and said arithmetic processing means, Means for executing the performance information processing program or the sound source processing program stored in the program storage means, and normally executes the performance information processing program to control the sound generation data on the data storage means at predetermined time intervals. Transfer control to the sound source processing program and execute it And means for executing the performance information processing program again after the end thereof, and, when executing the sound source processing program, program execution means for generating a sound signal in the modulation method due to the sound generation data on the data storage means; And a sound signal generating means for retaining the sound signal obtained by executing the sound source processing program and outputting the retained sound signal at regular output time intervals. do.

In the sound wave waveform generating apparatus according to the second state of the present invention, the program storage means, the address control means, the data storage means, the arithmetic processing means, and the program execution means have the same configuration as a general-purpose microprocessor, and no dedicated sound source circuit is required. Do not The sound signal output means is different from the general-purpose microprocessor, but is general in the category of sound wave waveform generators.

As a result, the circuit size of the entire acoustic wave generator can be significantly reduced, and even in the case of LSI, the manufacturing technology of the conventional microprocessor is the same, and the product ratio with respect to the raw material of the chip is also improved. It can be greatly reduced.

In addition, since the sound signal output means can be constituted by a simple latch circuit, the addition of this portion hardly increases the manufacturing cost.

In addition, when the modulation method is to be changed to, for example, the phase modulation method and the frequency modulation method, or the polyphonic number is to be changed, it is possible to cope by simply changing the sound source processing program stored in the program storage means. It is possible to greatly reduce the development cost of the acoustic wave generator, and to provide a new modulation scheme to the user, for example, by a ROM card.

The above operation is possible because the second state of the present invention realizes the following program architecture and data architecture.

That is, in the second state of the present invention, a data architecture for storing sound generation data necessary for generating sound by modulation method on the data storage means is realized. When the performance information processing program is realized, the sound generation data on the data storage means is controlled, and when the sound source processing program is executed, the sound signal is generated in the data storage means due to the sound generation data. In this way, data is communicated between the performance information processing program and the sound source processing program through the sound generation data on the data storage means, and access to the data storage means in each program is not related to the execution state of the other program. Since it is possible to do so, practically both programs can be made into an independent module structure, and a simple and efficient program structure can be achieved.

In addition to the data architecture, in the second state of the present invention, a performance information processing program is normally executed, for example, scanning of a keyboard key, various setting switches, demo playing control, and the like, for a predetermined time. A program architecture is realized in which sound source processing programs are executed at intervals, and when the processing is completed, the sound source processing program is returned to the performance information processing program. As a result, the sound source processing program may be forcibly interrupted to the performance information processing program due to an interrupt signal generated at a predetermined time interval in the interrupt control means, for example, so that the synchronization between the performance information program and the sound source processing program is performed. There is no need to take.

In addition, when the program executing means executes the sound source processing program, the processing time varies depending on the type of modulation method or the selectivity of the sound generation algorithm in the modulation method, but this change can be absorbed by the sound signal output means. Therefore, there is no need for a complicated timing control program for outputting the sound signal to the D / A converter or the like.

As described above, the data architecture of linking data between the performance information processing program and the sound source processing program through the sound generation data on the data storage means, and executing the sound source processing program at a predetermined time interval with respect to the performance information processing program. By realizing the program architecture and providing the sound signal output means, the sound source processing resulting from the efficient program control can be realized with the configuration almost the same as that of the general-purpose processor.

According to the third state of the present invention, a sound source processing according to a modulation method having an operator processing program for executing at least one operator process and an algorithm processing program for executing an algorithm process for determining an input-output relationship between each operator process A storage means for storing a program, a sound signal signal generating means for generating a sound signal by performing a batch operation independently of the operator processing resulting from the operator processing program and the algorithm processing resulting from the algorithm processing program, and the sound signal generating There is provided a sound wave generator, characterized by having a sound signal output means for outputting a sound signal generated by the means at a constant output time interval.

According to the sound wave generator according to the third state of the present invention, it is possible to realize a high sound source processing operating by a large amount of sound sound generation algorithms without using a dedicated sound source circuit, and also by the movement of the sound signal output means. Since the output rate of the sound signal is kept constant, it is possible to obtain a sound wave waveform without distortion.

According to the fourth state of the present invention, a performance information processing program for processing performance information and a sound source processing program by a modulation method for obtaining a sound signal, and at least one operator process which is a modulation processing unit are collectively executed Or program storage means for storing a sound source processing program having a processing structure in which algorithm processing as input / output processing between each operator processing is executed before execution thereof, address control means for controlling an address of the program storage means; Data storage means for storing sound generation data necessary for generating a sound signal by the modulation method, arithmetic processing means, and storing in the program memory means while controlling the address control means, the data storage means and the arithmetic processing means. The performance information processing program or the sound source A program for executing sound information processing on the data storage means, transferring control to the sound source processing program at predetermined time intervals, and executing it. Means for executing the performance information processing program again; program execution means for generating a sound signal by the modulation method due to the sound generation data on the data storage means when the sound source processing program is executed; There is provided a sound wave waveform generating device, comprising: a sound signal output means for holding the sound signal obtained by executing the sound source processing program and outputting the held sound signal at regular output time intervals.

In the sound wave waveform generating apparatus according to the fourth state of the present invention, as an architecture of a sound source processing program, an algorithm that is each input / output process between each operator process after at least one operator process, which is a modulation processing unit, is executed collectively or before its execution. It has a processing structure in which processes are executed in a batch. Usually, if one operator process is completed, the algorithm has a processing structure that determines the input / output relation to the next operator by the specified algorithm and performs the next operator process. You must be willing to listen to some kind of sound processing program. On the other hand, in the acoustic wave generator according to the fourth state of the present invention, even when the sound source processing is performed by selecting among a plurality of algorithms in the modulation method, a plurality of types of algorithm processing parts can be used, and if necessary, they are switched. As a result, the sound source processing program can be made very compact.

[Summary of this embodiment]

First, the outline | summary of this embodiment is demonstrated.

FIG. 1 is an overall configuration diagram of the present embodiment, wherein the external memory 116 is constituted by one chip, and two CPUs (central operation controllers) of a master and a slave thereof exchange information with each other. In doing so, the sound source processing for music production is shared.

For example, in the case of 16-channel polyphonic, eight channels are processed by the master CPU 101, and the remaining eight channels are processed by the slave CPU 102.

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

This sound source method is automatically assigned to a specific instrument sound, for example trumpet or tuba.However, the sound source color can be selected using the selection switch for other instrument sounds. It can be selected automatically according to the playing strength.

In addition, for example, it is possible to assign two channels at the same time with a single taper, and another sound source method, for example, an attack part is a PCM method, and a sustain part is an FM method.

For example, in the case of the FM system, processing is performed by software in accordance with an algorithm of sound source processing, which takes too long when performed with a general-purpose CPU.

The above is the outline of this embodiment.

[Configuration of this embodiment]

EMBODIMENT OF THE INVENTION Hereinafter, the Example of this invention is described, referring drawings.

In Fig. 1, first, the external memory 116 has a sound control parameter for the target values of envelope values, a sound wave waveform in the PCM (pulse code modulation) method, or a sound wave waveform in the D PCM (differential pulse code modulation) method. Etc. are remembered.

On the other hand, the master CPU (hereinafter abbreviated as M CPU) 101 and slave CPU (hereinafter abbreviated as S CPU) 102 are accessed, shared, and sound source processing. Since these CPUs share waveform data and the like of the external memory 116 together, there is a possibility that competition may occur when data is read from the external memory 116 in this state. As such, each of the M CPU 101 and the S CPU 102 avoids the contention access for access through the M CPU external memory access address latch 103 and the S CPU external memory access address latch 104. As the circuit 105 outputs the address signal for external memory access and the external memory control data from the output terminals 111 and 112, a competition between an address in the M CPU 101 and an address in the S CPU 102 is eliminated. Can be avoided.

The data read out from the external memory 116 due to the addressing is input to the external memory selector unit 106 at the terminal 115 which is external memory data. The external memory selector unit 106 passes the read data through the data bus MD and the data input to the M CPU 101 due to the control signal from the access address avoidance circuit 105, and passes through the data bus SD. The data is separated into data input to the S CPU 102 and input to the M CPU 101 and the S CPU 102, respectively. This can also avoid data competition.

Then, for each data, after the sound source processing is performed by the M CPU 101 and the S CPU 102 by software, all the pronunciation channels are accumulated, and the left output of the Left D / A converter 107 is obtained. The left and right channel left and right analog outputs and the right and right analog output signals are output from the terminal 113 and the right output terminal 114 of the Right D / A converter section 108 as sound signals.

Next, FIG. 2 is a block diagram showing the internal structure of the M CPU 101. As shown in FIG.

In the figure, a program for sound control described later is stored in the control ROM 201, and the ROM address control unit 205 sequentially outputs a program word (command) having a designated address through the ROM address decoder 202. FIG. Specifically, the spelling number of each program word is 28 bits, for example, and a next address method in which a part of the program word is input to the ROM address control unit 205 as a lower part of the address to be read next (in the page). It is. As a matter of course, it may be constituted by a CPU of a normal program counter system.

The command analysis unit 207 interprets the OP code of the command to be output from the control ROM 202, and sends a control signal to each part of the circuit for executing the designated operation.

The RAM address control unit 204 designates an address of a corresponding register in the RAM 206 when an operand of an instruction in the control ROM 201 specifies a register.

In the RAM 206, various sound control data described later in FIG. 11, FIG. 12, etc. are stored for eight sounding channels, and various buffers described later are stored and used for the sound source processing described later.

The ALU unit 208 and the multiplier 209 are based on the instructions in the command analysis unit 207 when the instruction in the control ROM 31 is an operation instruction, so that the former performs addition and subtraction and logical operation, and the latter performs multiplication. do.

The interrupt control unit 203 causes the reset release signal A, the ROM address control unit 205 and the respective D in FIG. 1 to the S CPU 102 of FIG. The interrupt signal is supplied to the / A converters 107 and 108.

In addition to the above-described phrase, the M CPU 101 shown in FIG. 2 is provided with interfaces relating to the following various buses.

That is, the interface 215 of the address bus MA for specifying the address of the memory for accessing the external memory 116, and the accessed data can be transferred between the M CPU 101 through the external memory selector 106. The interface 216 of the data bus MD to be used, the interface 212 of the bus Ma which designates the address of the RAM inside the S CPU 102 to execute data transfer with the S CPU 202, and the M CPU 101 Interface 213 of data bus Dout for storing data with CPU 102, interface 214 which is data bus Din for M CPU 101 to read data from S CPU 102, Left D / A conversion With the base 107, the interface 217 of the D / A data transmission bus for transmitting the final output waveform to the Right D / A converter section 108, and an external switch or keyboard section (see FIG. 8). There are input / output ports 210 and 211 for carrying data between them.

Next, the internal structure of the S CPU 102 is shown in FIG.

Since the S CPU 102 only performs the sound source processing in response to the processing start signal from the M CPU 101, the S CPU 102 performs an operation with the interrupt control unit corresponding to FIG. 2 203 and the external circuits corresponding to FIGS. 210 and 211 in FIG. There is no interface for outputting a sound signal to the input / output port for receiving data and the left D / A converter section 107 and the Right D / A converter section 108 corresponding to FIG. Other circuits of 301, 302, and 304-309 have functions similar to those of 201, 202, and 204-209 of FIG. In addition, each of the interfaces 303 and 310-313 is formed opposite to each other in Fig. 212-216. In addition, an internal RAM address of the S CPU 102 designated by the M CPU 101 via the bus Ma is input to the RAM address control unit 304, and a corresponding address is designated for the RAM 306. FIG. Thereby, for example, accumulated waveform data for the pronunciation channel generated in the S CPU 102 and held in the RAM 306 is output to the M CPU 101 via the data bus DIN. This will be described later.

In addition to the configuration shown above, in this embodiment, the function key 801 and the keyboard key 802 shown in FIG. 8 are connected through the input port 210 of the M CPU 101. These parts constitute a substantial musical instrument operation section.

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

FIG. 6 (b) shows the internal structure of the D / A converter sections 107 and 108 (the contents of both converter sections are the same) of Left and Right of FIG. 1, and shows one sample of the sound produced by sound source processing through the data bus. Data is input to the latch 601. Then, when the sound source processing end signal is inputted to the clock force of the latch 601 from the command analysis unit 207 (FIG. 2) of the M CPU 101, the sound data for one sample on the data bus is shown in FIG. As shown in the drawing, the latch 601 latches the latch 601.

Here, the time required for the sound source processing described above is converted by the software for sound source processing. Therefore, when the sound source systems are different, the timing at which each sound source process ends and the sound data is latched in the latch 601 is not constant. Therefore, the output of the latch 601 cannot be input directly to the D / A converter 603 as shown in FIG. 6 (a).

In this embodiment, as shown in FIG. 6 (b), the output of the latch 601 is latched back to the latch 602, and the sound signal is received by the same interrupt signal at the sampling claw interval output from the interrupt control unit 203. FIG. The latch 602 is latched and output to the D / A converter 603 at a predetermined interval.

Since two latches are used to absorb the change in processing time by the sound source method, a complicated timing control program for outputting the sound data to the D / A converter becomes unnecessary.

[Overall operation of this embodiment]

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

This embodiment basically operates with the M CPU 101 as the center, and as shown in the main flowchart of Fig. 4A, a series of processes of S402 to S410 are repeatedly performed. The actual sound source processing is performed by interrupt (INTERRUPT) processing. The actual sound source processing is performed by interrupt (INTERRUPT) processing. Specifically, the M CPU 101 and the S CPU 102 are interrupted at any given time, and each CPU performs sound source processing to produce eight channels of sound. After the processing, the sound waveforms for the eight channels and the 16 channels of each CPU are added and output from the Left D / A converter section 107 and the Right D / A converter section 108. After that, it returns to the main flow from the interrupt state. In addition, the interrupt mentioned above is performed periodically due to the hard timer in the interrupt control unit 203 of FIG. This period is equivalent to the sampling period at the time of sound output.

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

In the main flowchart of FIG. 4 (a), if an interrupt is interrupted in the interrupt control unit 203 while the processing of S402-S410 is repeatedly executed, the M CPU interrupt processing of FIG. (B) and S of FIG. Two processes of CPU interrupt processing are started simultaneously. The "sound source processing" of the same drawing (b) and (c) is shown in the drawing (d).

By the way, the main flowchart of FIG. 4 (a) is executed by the M CPU 101 in the state where interrupt is not interrupted by the interrupt control unit 203. The flow of processing other than the sound source processing is shown.

First, the power supply is turned on, and initial settings such as the contents of the RAM 206 of the M CPU 101 are performed (S401).

Next, a function key connected to the outside of the M CPU 101, for example, a tone switch shown in FIG. 27, is scanned (S402), and the state of each switch is inputted from the input port 210 to the RAM 206. It is picked up in the inner key buffer area. As a result of the scanning, the state-changed function key is identified, and processing of the corresponding function is performed (S403). For example, when a rhythm performance is attached to a set of musical number numbers, an envelope number, and an additional function, a set of rhythm numbers is performed.

Thereafter, the key being pushed is collected in the same manner as in the case of the function key (S404), and keying processing is performed as the changed key is identified (S405).

Next, when the demo playing key is pressed by the function key 801 (see FIG. 8), the demo playing data (sequencer data) is sequentially read from the external memory unit 116, and key assignment processing is performed (S406). ). When the rhythm start key is pressed, the rhythm data is sequentially read from the external memory 116, and keying processing or the like is performed (S407).

Thereafter, the timer process described below is performed (S408). That is, the time value of the time data incremented by the interrupt timer process (S412) described later is determined, and compared with the time control sequencer data read out sequentially for demo performance control or the rhythm data for time control read out for the rhythm performance control. As a result, time control is performed when demo performance of S406 or rhythm performance of S407 is performed.

In the pronunciation process S409, an envelope is added to the pitch of the sound to be processed, and a pitch envelope process is performed on the corresponding pronunciation channel.

In addition, a flow preparation process is executed (S410). In this processing, processing such as changing the state of the sound tone channel of the note number at the start of the keyboard to the dry key or changing the state of the sounding channel of the note number made in the noise during noise in S405 is performed. Is done.

Next, FIG. 4 (b) illustrates the M CPU interrupt processing.

When the M CPU 101 is interrupted by the interrupt control unit 203, which is the M CPU 101, the processing of the main flowchart of FIG. 4 (a) is interrupted, and the M CPU interrupt processing of FIG. 4 (b) is executed. This is disclosed. In this case, in the program of the M CPU interrupt processing, it is controlled so that the contents of the register or the like, which are stored in the program of the main flow in Fig. 4A, are not rewritten. This eliminates the need for retrieval and return of registers that are performed at the start and end of normal interrupt processing. As a result, the transition between the processing of the main flowchart of FIG. 4A and the M CPU interrupt processing is performed quickly.

Subsequently, the sound source processing is started in the M CPU interrupt scene processing (S411). This sound source processing is shown in FIG. 4 (b).

Simultaneously with the above operation, the S CPU reset release signal A (see FIG. 1) is output from the interrupt control unit 203 of the M CPU 101 to the ROM address control unit 305 of the S CPU 102. Execution of the S CPU interrupt processing of FIG. 4C is started in the CPU 102. FIG.

At the same time as the sound source process (S411) in the M CPU interrupt process, the sound source process is started in the S CPU interrupt process (S415). In this manner, both the M CPU 101 and the S CPU 102 simultaneously execute sound source processing for each of the eight sounding channels, so that the sound source processing for the 16 sounding channels can be performed in the processing time of the eight sounding channels, and the processing speed is approximately doubled. (The above will be described later with reference to FIG. 5).

Subsequently, the M CPU 101 waits for the end signal of the S CPU interrupt processing to come from the S CPU 102 after the S412 interrupt timer processing (S413). In addition, in the interrupt timer process, the interrupt process shown in FIG. 4 (b) is executed every fixed sampling period, and the value of the time data, not specifically shown, on the RAM 206 (FIG. 2) is incremented. . In other words, the value of this time data shows the time elapsed. The time data obtained in this manner is used for time control in the timer process S408 of the main flow of FIG. 4A as described above.

When the sound source processing in step S415 is finished in the S CPU interrupt processing in FIG. 4 (c), the S CPU 102 sends the S CPU to the ROM address control unit 205 of the M CPU 101 by the command analysis unit 307 of the S CPU 102. The processing end signal B (see FIG. 1) is input. As a result, the determination in step S413 becomes YES in the M CPU interrupt processing of FIG. 4 (b).

As a result, the waveform data produced by the S CPU 102 via the data bus Din in FIG. 1 is read into the RAM 206 of the M CPU 101 (S414).

In this case, since the waveform data is stored in a predetermined buffer area (buffer B described later) on the RAM 306 of the S CPU 102, the command analysis unit 207 of the M CPU 101 designates the S CPU internal address. Waveform data is read by designating the buffer address to the RAM address control unit 304 via the bus Ma.

In S415 ', the contents of the buffer area are latched by the latch 601 (see FIG. 6) of the Left D / A converter section 107 and the Right D / A converter section 108. As shown in FIG.

Next, the operation of the sound source processing executed in step S411 of the M CPU interrupt processing or step S415 of the S CPU interrupt processing will be described using the flowchart shown in FIG. 4 (d).

First, the area for adding waveform data of the RAM 206 or the RAM 306 is cleared (S416).

Next, sound source processing is performed for each channel of the pronunciation channel (S417-S424), and finally, waveform data obtained by adding eight channels to the predetermined buffer area B is obtained at the end of the eighth channel sound source processing. These detailed processes will be described later.

Next, FIG. 5 is a flowchart conceptually showing the processing relationship of the flowcharts of FIGS. 4 (a), (b), and (c) described later, in which the M CPU 101 and the S CPU 102 respectively perform sound source processing. It shows how to share.

First, any process A (hereinafter, B, C, D, E, F are also the same) is performed (S501). This " processing " corresponds to, for example, " function key processing " or " keyboard processing " in the main flowchart of FIG. 4 (a). After that, the processing enters the M CPU interrupt processing and the S CPU interrupt processing, and simultaneously the sound source processing by the M CPU 101 and the S CPU 102 is started (S502, S503). Then, at the end of the S CPU interrupt processing in the S CPU 102, the S CPU processing end signal B is input to the M CPU 101. In the M CPU interrupt processing, when the sound source processing ends faster than the S CPU interrupt processing, and when the S CPU processing end B is identified in the S CPU interrupt processing, the waveform data generated by the S CPU 102 is sent to the M CPU 1010, and the M CPU interrupt processing is performed. The waveform data generated by the CPU 101 is arranged together and output to the Left D / A converter 107 and the Right D / A converter 108. Then, the process returns to a certain process B of the main flowchart.

The operation described above is repeated while the sound source processing is performed for all the pronunciation channels (16 channels in which the M CPU 101 and the S CPU 102 are combined in this embodiment) (S504-S516). This repetition process is continued during pronunciation of the musical sound.

[Data Structure in Sound Source Processing]

Next, specific examples of the sound source processing executed in S411 of FIG. 4 (b) and S415 of FIG. (C) will be described.

In the present embodiment, both CPUs of the M CPU 101 and the S CPU 102 share the sound source processing for each of the eight channels. As shown in FIG. 9, the data for the sound source processing for these eight channels is set in the region for each sound channel in each of the RAMs 206 and 306 of the M CPU 101 and the S CPU 102.

Moreover, as shown in FIG. 12, each of the buffers BF, BT, B, and M is secured in this RAM.

In this case, each sound source method can be set in each pronunciation channel region of FIG. 9 by the operation described later, as shown in FIG. 10, and if the sound source method is set, as shown in FIG. In the same data format of each sound source system, data is set in each region of each pronunciation channel in FIG. In addition, in the present embodiment, as described later, it is possible to assign a sound source method for each pronunciation channel.

In Table 1 showing the data format of each sound source system in Fig. 11, G is the sound source system No. which is a number for identifying the sound source system. The next A indicates an address specified when waveform data is read during sound source processing, where A _I , A ₁ and A ₂ are integer portions of the current address, and waveform data of the external memory 116 (FIG. 1) It corresponds directly to the address stored. A _F is a fractional part of the old address and is used for interpolation of waveform data read from the external memory 116.

Next, A _E represents an end address and A _L represents a loop address, respectively. Next, P _I , P _1, and P ₂ denote integer portions of pitch data, and P _F denote fractional portions of pitch data. For example, P _I = 1, P _F = O is the pitch of the sound source, P _I = 2, P _F = 0 is the pitch of one octave, and P _I = O, P = 0.5 is the pitch of one octave. Represent each.

Next X _P represents the previous sample data, and X _N represents the storage of the next sample data. D denotes a difference value of the size between two adjacent sample data, and E denotes an anbily-loof value. In addition, 0 is an output value, and C, which will be described later, indicates a flag used when changing the sound source system to be assigned to the pronunciation channel by the performance information.

Various other control data will be described when describing each sound source method to be described later.

In this manner, when the data shown in FIG. 11 is secured in the RAMs 206 and 306 of the M CPU 101 and the S CPU 102, and the sound source method described below is determined, the respective channels shown in FIG. Each time, data is set in the format shown in FIG.

Hereinafter, the sound source processing of each sound source system executed by using the data structure will be described in sequence. In addition, these sound source processes are executed by the command analysis unit 207 or 307 of the M CPU 101 or the S CPU 102 for the sound source processing program stored in the control ROM 201 or 301. This is realized by executing the analysis. Hereinafter, unless otherwise indicated, the processing is performed based on this premise.

First, in the flowchart of FIG. 4 (d), when each sound source process (somewhere in S417 to S424) for each channel is entered, it is shown in FIG. 11 stored in the corresponding pronunciation channel of the RAMs 206 and 306. FIG. The sound source method No. is determined from among the data of the data format (Table 1) to be performed, thereby determining whether the sound source method of the sound source method described below is executed.

[Sound source processing by PCM method]

When the sound source method No. indicates the PCM method, sound source processing by the PCM method shown in the operation flowchart of FIG. 13 below is executed. Each variable in the flow is each data of the PCM format shown in Table 1 of FIG. 11 stored in the pronunciation channel region of FIG. 9 on the RAMs 206 and 306 of the M CPU 101 or the S CPU 102.

Fig. 15 (a) shows an address in which waveform data as the current processing target is stored among the addresses in which PCM waveform data on the external memory 116 (Fig. 1) is stored (A _I , A _F )

First, the pitch data (P _I, P _F) is added to the current address (S1301). This pitch data corresponds to the kind of key that was press-dried in the key 801 of FIG.

Then, it is determined whether or not the integer part A _I of the added address is changed (S1302). If the determination is No, the arithmetic processing is D x A _F using the difference value D which is the difference between the respective sample data X _E and X _{F in} the address A _I +1 and A _I in Fig. 15 (a). The interpolation data value 0 corresponding to the fractional part A _F of the address is calculated (S1307). The difference value D is obtained by sound source processing at the interrupt timing before this time (see S1306 to be described later).

Then, the sampling data X _F corresponding to the integer part A _I of the address is added to the interpolation data value 0, and new sample data 0 corresponding to the current addresses A _I and A _F (X _{Q in} FIG. 15 (a)). (Corresponds to Equation 2)) (S1308).

Then, the envelope value E is elevated to this sample data (S1309), and the content of the obtained zero is the waveform data buffer B (the twelfth) in the RAM 206 or 306 of the M CPU 101 or the S CPU 102. (See FIG. 1) (S1310).

Thereafter, the flow returns to the main flow of FIG. 4 (a), interrupts in the next sampling period, and the operation flowchart of the sound source processing in FIG. 13 is executed again, and the pitch data P is applied to the current addresses A _I and A _{F. I} and P _F are added (S1301).

The above operation is repeated until the integer part A _I of the address is replaced (S1302).

In the meantime, the sample data X _F and the difference value D are not updated, only the interpolation data O is updated in accordance with the address A _F , and sample data X _Q is obtained each time.

Next, when the pitch data P _I and P _F are added to the current addresses A _I and A _F in S1301, and the integer part A _I of the current address is changed (S1302), the address A _I is the end address A _F. It is determined whether or not it has reached or exceeded (S1303).

If the determination is YES, the next root process is performed. In other words, the address A is added to the end address (A _I -A _E), the loop _L of the address A as an _E beyond, the loop reproduction in the integer part of A _I the new current address is initiated obtained (S1304). The end address A _E is an address on the external memory 116 (FIG. 1) in which the last waveform sample data of the PCM waveform data is stored. The loop address A _L is an address of a position at which the player wants to repeat the output of the waveform, and the loop operation is realized in the PCM method by the above operation.

If the determination of S1303 is NO, the processing of S1304 is not executed.

Next, the sample data is updated. Here, each sample data corresponding to the newly updated current address A _I and the address immediately preceding one (A _I -1) in the external memory 116 (FIG. 1) is read as X _N and N _P respectively (S1305). .

Further, the difference value thus far is updated to the difference value D between X _N and X _P updated (S1306).

The operation after this is as described above.

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

[Sound source processing by DPCM method]

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

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

In the figure, the sample data X _P corresponding to the address A _I of the external memory 116 (FIG. 1) corresponds to an address A _I -1 not specifically shown before one of the addresses A _I. The value is obtained from the difference of sample data.

Since the difference D from the next sample data is stored in the address A _I of the external memory 116 (FIG. 1), the sample data of the next address is obtained by X _P + D, which is the new sample data X _P. Is moved as

In this case, as shown in the figure, if A _F is set, the sample data corresponding to the current address A _F is obtained as X _F + D × A _F.

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

By adopting such a DPCM method, when quantizing a waveform such as a voice or a sound with a small difference between adjacent samples, the quantization can be performed with a much smaller number of bits than in a conventional PCM method. It is clear.

The operation of the DPCM method described above will be described using the operation flowchart of FIG. Each variable in the flow is each data of the DPCM format shown in Table 1 of FIG. 11 stored in the pronunciation channel region of FIG. 9 on the RAMs 206 and 306 of the MCPU 101 or the SCPU 102.

External memory 116 (FIG. 1) of an address which is stored the DPCM difference waveform data on, showing the addresses in the current data to be processed in the stored in the second 15 (b) Fig (A _I, A _F )

First, the current address (A _I, A _F) the pitch data (P _{I, F} A) to be added (S1401).

Then, it is determined whether or not there is a change in the integer part A _I of the added address (S1402). If the determination is NO, 0 of the interpolation data value corresponding to the fractional part A _F of the address is calculated by the arithmetic processing of D × A _F using the difference value D in the address A _I of FIG. 15 (b) (S1414). The difference value D is obtained by sound source processing at the interrupt timing before this time (see S1406 and S1410 described later).

Next, sample data X _F corresponding to the even number A _I of the address is added to the interpolation data value 0, and new sample data 0 corresponding to the current addresses A _I and F _F (X in FIG. 5 (b)). Corresponding to _Q ) is obtained (S1415).

Subsequently, the envelope data E is multiplied by this sample data (S1416), and the content of 0 obtained is the waveform data buffer B in the RAM 206 or 306 of the MCPU 101 or SCPU 012 (see FIG. 12). ) Is added (S1417).

Thereafter, the flow returns to the main flow of FIG. 4 (a), interrupts in the next sampling period, and the operation flowchart of the sound source processing of FIG. 14 is executed again, and the pitch data (at the current address A _I , A _F ) is executed. P _I and P _F are added (S1401).

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

During this time, the sample data X _P and the difference value D are not updated, and only interpolation data 0 is updated in accordance with the address A _F , and new sample data X _Q is obtained each time.

Next, if the integer part A _I of the current address is changed as a result of adding the pitch data P _I and P _F to the current addresses A _I and A _F in S1410 (S1402), the address A _I is the end address A _E. It is determined whether it has reached or exceeded (S1403).

If the determination is NO, the sample data corresponding to the integer part A _I of the current address is calculated by the loop processing of S1404-S1407 below. In other words, the value before the constant portion A _I of the current address is stored in the variable A _I (see the column of DPCM in Table 1 in FIG. 11) before. This, as cement sequence of cleaners in the later-described S1406 previous A _I uigap the difference waveform data on the external memory 116 (FIG. 1) while sequentially in Cleveland cement in S1406 is specified by the previous A7 _P in S1407 from S1407 is It is read as D and accumulated in the sequential sample data X _P in S1405. When the value of the previous A _I becomes equal to the integer portion A _I of the current address after the change, the value of the sample data X _P becomes a value corresponding to the integer portion A _I of the current address after the change.

In this way, when the sample data X _P corresponding to the integer part A _I of the current address is obtained, the determination of S1404 is YES, and the processing proceeds to the above-described calculation processing of interpolation value (S1414).

The above-described sound source processing is repeated for each interrupt timing, and if the determination of S1403 has changed to YES, the process proceeds to the next loop process.

First, as many addresses A _I- A _E as having exceeded the end address A _E are added to the loop address A _L and the obtained address becomes the integer part A _I of the new current address (S1408).

By the number of times the address advances in the loop address A _L , the operation of accumulating the difference value D is repeated several times, so that the sample data X _P corresponding to the integer part A _I of the new current address is calculated. In other words, first, the sample data X _P as the initial settings are preset and the value of the sample data X _PL (see of the DPCM of 11 degrees Table 1 column) in the loop the address A _L, yet prior to this A _I, the loop address A _It becomes the value of _L (S1409). And the following process of S1410-S1413 is repeated. That is, while the value of the previous A _I is sequentially incremented in S1413, the differential waveform data on the external memory 116 (FIG. 1) indicated by the previous A _I in S1410 is read as D, and the sequential sample data in S1412. Accumulate in X _P. When the value of the previous A _I becomes equal to the integer portion A _I of the new current address, the value of the sample data X _P becomes a value corresponding to the integer portion A _I of the new current address after the loop processing.

In this way, when the sample data X _P corresponding to the integer part A _I of the new current address is obtained, the determination of S1411 is YES, and the processing proceeds to the above-described interpolation value calculation processing (S1414).

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

[Sound Source Processing by FM Modulation Method (Part 1)]

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

In the FM modulation system, hardware or software of the same content, commonly referred to as an operator represented by OOP1-OP4 in FIG. 18, is used, and the generation of the musical sound in accordance with the interconnection in the connection rule where they appear as Algorithms 1-4 in FIG. Is done. In this embodiment, the FM modulation method is implemented by software.

Next, the operation of the first embodiment when the sound source processing is performed by the two operators using the operation flowchart of the sixteenth (a) will be described. The algorithm of processing is shown in Fig. (B). Each variable in the flow is the data of the FM format shown in Table 1 of FIG. 11 stored in the pronunciation channel region of FIG. 9 on the RAMs 206 and 306 of the MCPU 101 or SCPU 102. FIG.

First, the process of operator 2 (OP 2) which is a modulator is performed.

As for the pitch processing, no interpolation is performed as in the PCM method, and only the constant address A ₂ is used. In other words, it is assumed that the modulated waveform data is stored in the external memory 116 (FIG. 1) at a very small interpolation interval.

First, pitch data P ₂ is added to the current address A ₂ (S1601).

Next, the feedback output F _O2 is added to this address A ₂ as a modulation input, and a new address A _M is obtained (S1602). The feedback output F _O2 is obtained by executing the processing of S1605 to be described later at the last interrupt timing.

In addition, the value of the sine wave corresponding to the address A _M2 (phase) is calculated. In fact, sine wave data is stored in the external memory 116 (FIG. 1), and is obtained by moving the sine wave data to a table at the address A _M2 (1603).

Subsequently, the envelope rope value E ₂ is multiplied by the sinusoidal wave data to obtain an output O ₂ (S1604).

Subsequently, the feedback output level F _L2 is multiplied by this output O ₂ to obtain a feedback output F _O2 (S1605).

This output F _O2 becomes the input of the operator 2 (OP2) at the next interrupt timing in the present embodiment.

The modulation.level M _L2 is multiplied by O ₂ to obtain a modulation output M _O2 (S1606).

The modulation output _O2 M is the modulation input to the operator 1 (OP1).

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

First, the pitch data P ₁ is added to the current address A ₁ of the operator 1 (S1607). The modulation output M _O2 described above is added to this value to obtain a new address A _M1 (S1608).

Next, the value of the sine wave corresponding to this address A _M1 (phase) is read from the external memory 116 (FIG. 1) (S1609), and the envelope value E ₁ is multiplied by this to output the sound wave waveform O ₁ . It is obtained (S1601).

This is added to the buffer B (see FIG. 12) in the RAM 206 (FIG. 2) or 306 (FIG. 12) (S1611), and the FM modulation process for one sound channel is completed.

[Sound source method (TM) by TM modulation method]

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

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

The above-described FM modulation method,

e = A.Sin {ωct + I (t) .Sin ωmt}

Is based on the expression. Where ωct is the carrier phase angle (carrier signal), Sin ωmt is the modulation wave phase angle (modulation signal), and I (t) is the modulation index.

On the other hand, in the present embodiment, the phase modulation method called TM modulation method,

e = Af _T {fc (t) + I (t) .Sin ωmt}

Is based on the expression. Here, fT (t) is a triangular wave number, and is defined as the following number for each phase angle region (where ω is input).

f _T (ω) = 2 / πω

‥ (Area: 0 ω π / 2)

f _T (ω) =-1 + 2 / π (3π / 2-ω)

‥ (Area: π / 2 ω 3π / 2)

f _T (ω) =-1 + 2 / π (ω-3π / 2)

‥ (Area: 3π / 2 ω 2π)

Fc is called a modified sine wave and is a carrier signal generation number obtained by accessing an external memory 116 (FIG. 1) stored in different sine wave data for each phase angle region at a carrier phase angle ωct. Fc for each phase angle region is defined as follows.

f _C (t) = π / 2 Sin ωct

‥ (Area: 0 ωt π / 2)

f _C (t) = π-π / 2 Sin ωct

‥ (Area: 0 ωt 3π / 2)

f _C (t) = 2π + π / 2 Sin ωct

‥ (Area: 3π 2ω t _C π / 2π)

(Where n is an integer)

In the TM modulation method, the above-described triangular wave number is modulated by the addition signal obtained by adding the modulation signal sinωmt to the carrier signal generated at the irrigation water f _C (t) at the rate indicated by the modulation index I (t). As a result, a sine wave can be generated when the value of the modulation index I (t) is 0, and a very deeply modulated waveform can be generated by increasing the value of I (t).

Here, various signals can be used instead of the modulated signal sinωmt, and as described below, the operator's output of the previous operation can be returned to a constant feedback level, or the output of another operator can be input. .

The sound source processing by the TM modulation method of this principle is explained using the operation flowchart of FIG. 17 (a). This case is also an example of a case where sound source processing is performed by two operators as in the FM modulation method of FIG. 16, and the algorithm of the processing is shown in FIG. 17 (b). Each variable in the flow is the data of the TM format shown in Table 1 of FIG. 11 stored in the pronunciation channel region somewhere in FIG. 9 on the RAMs 206 and 306 of the MCPU 101 or the S CPU 102. .

First, processing of operator 2 (OP2), which is modulation, is performed. Interpolation is not performed with respect to the pitch processing as in the PCM method, only the constant address A ₂ .

First, pitch data P ₂ is added to the current address A ₂ (S1701).

Next, the modified sine wave corresponding to the address A ₂ (phase) is read from the external memory 116 (FIG. 1) by the modified sine transform fc, and a carrier signal is generated as O ₂ (S1702).

Then, the blood deukbaek output _O2 F (S1706) as a modulation signal to the above-mentioned O ₂ of the carrier signal are added, a new address is obtained is a O ₂ (S1703). Feedback output _O2 F (S1706) is added, the new address is obtained is a O ₂ (S1703). The gain-back output F _O2 is obtained by performing the processing of S1706 which will be described later because of the previous interrupt timing.

Then, the value of the triangular waveform corresponding to the above-mentioned O ₂ adds the address is calculated. In reality, the above-described triangular wave data is stored in the external memory 116 (FIG. 1), and is obtained by moving the triangular wave data to a table at the address O ₂ (S1704).

Subsequently, the triangle wave data is multiplied by an envelope value E ₂ to obtain an output O ₂ (S1705).

Subsequently, the feedback level F _L2 is multiplied by this output O ₂ to obtain a feedback output F _O2 (S1707).

This output F _O2 becomes an input to operator 2 (OP2) at the next interrupt timing in the present embodiment.

The modulation.level M _L2 is multiplied by O ₂ to obtain a modulation output M _O2 (S1707). The modulation output _O2 M is the modulation input to the operator 1 (OP1).

Next, the process moves to the operator OP1. This processing is almost the same as that of the operator 2 described above except that there is no modulation input by the feedback output.

First, the pitch data P ₂ is added to the current address A ₁ of the operator 1 (S1708), and the above-described transformed sine conversion is performed on the obtained value to obtain a carrier signal as O ₁ (S1709).

Next, a modulation output described in the O ₁ M _O2 are added and a new O ₁ (S1710), the value O is ₂ the triangular wave conversion (S1711), In addition, the envelope value E is _first multiplied tone waveform output O ₁ is obtained (S1712).

This is added to the buffer B (see FIG. 12) in the RAM 206 (FIG. 2) or 306 (FIG. 12) (S1713), and the TM modulation process for one sound channel is completed.

In the above, sound source processing by four methods, PCM, DPCM, FM, and TM, was demonstrated. Of these, the two methods of FM and TM are modulation methods. In the above-mentioned examples, the processing is performed by two operators resulting from the algorithm shown in Figs. 16 (b) and 17 (b). The process has a larger number of operators, and the algorithm is more complicated. An example is shown in FIG. In Algorithm 1 of the figure, four layers of modulation are performed including the feedback input, thereby obtaining a complicated waveform. In Algorithm 2, two sets of algorithms having a feedback input are arranged in parallel, and are suitable for expressing a change in tone when moving from attack to sustain, for example. Algorithm 4 has a feature close to that of sine wave synthesis.

Examples of the conventional sound source processing of the FM system and the TM system by the four-operator of FIG. 18 will be described in sequence using FIG. 19 and FIG.

[1 (2) by FM modulation method]

FIG. 19 is an operation flowchart of a conventional sound source processing resulting from the FM modulation system corresponding to Algorithm 1 of FIG. Each variable in the flow is stored in the pronunciation channel region somewhere in FIG. 9 on the RAMs 206 and 306 of the M CPU 101 or the S CPU 102. In addition, although it does not correspond to each data of the FM format of Table 1 of FIG. 11, it is clear that it can be easily achieved by extending the data format.

First, the pitch data P ₄ is added to the current address A ₄ of the operator 4 (OP4) (S1901).

Next, the feedback output F _O4 (S1905) is added to this address A ₄ as a modulation input, and a new address A _M4 is obtained (S1902). In addition, the value of the sine wave corresponding to the address A _M4 (phase) is read from the external memory 116 (see FIG. 1) (S1908), and the envelope value E ₄ is multiplied by this to obtain an output O ₄ . (S1904). Thereafter, the feedback. Level F _L4 is multiplied to the output O ₄ to obtain a feedback output F _O4 (S1905). The modulation.level M _L4 is multiplied by the output O ₄ to obtain a modulation output M _O4 (S1906). This modulation output _MO4 is a modulation input to the next operator 3 (OP3).

Next, the process moves to operator 3 (OP3). This process is almost the same as that of the operator 4 described above, except that there is no modulation input by the feedback output. First, pitch data P ₃ is added to the current address A ₃ of operator 3 (OP3) (S1907). Next, the modulation output M _O4 is added to the address A ₃ as a modulation input, and a new address A _M3 is obtained (S1908). In addition, the value of the sine wave corresponding to this address A _M3 (phase) is read from the external memory 116 (FIG. 1) (S1909), and the envelope value E ₃ is multiplied to obtain an output O ₃ (S1910). ). Then, the modulation Level _L3 M is multiplied to the output O ₃ are the modulated output M _O3 obtained (S1911). This modulation output _MO3 is a modulation input to the next operator 2 (OP2).

Next, although the processing of the operator 2 (OP2) is executed, this processing is the same as in the case of the operator 3 described above because only the modulation inputs are different, and a description thereof is omitted.

Finally, the process of the operator OP1 is entered, but this process is also performed up to step S1920 as before. The sound waveform output O ₁ obtained in S1920 is accumulated in the buffer B (see FIG. 12) as a carrier (S1921).

[Sound source processing by TM modulation method (Part 2)]

FIG. 20 is an operation flowchart of a conventional sound source processing resulting from the TM modulation system corresponding to Algorithm 1 of FIG. Each variable in the flow is stored in the pronunciation channel region of FIG. 9 on the RAMs 206 and 306 of the M CPU 101 or the S CPU 102. In addition, although it does not correspond to each data of TM format of Table 1 of FIG. 11, it is clear that it can be easily achieved by extending the data format of FIG.

First, the pitch data P ₄ is added to the current address A ₄ of the operator 4 (OP4) (S2001).

Next, the modified sine wave corresponding to the above-described address A ₄ (phase) is read from the external memory 116 (see FIG. 1) by the modified sine change fc, and a carrier signal is generated as O ₄ (S2002). . The feedback output F _O4 (see S2007) is added to the output O ₄ as a modulation signal, and a new address O ₄ is obtained (S2003). Next, the value of the B triangle wave corresponding to this address O ₄ (phase) is read from the external memory 116 (FIG. 1) (S2004), and the envelope value E ₄ is multiplied by this to obtain the output O ₄ . (S2005). Thereafter, modulation to this output O ₄ . The level M _L4 is multiplied to obtain a modulation output M _O4 (S2006), and the output O ₄ is fed back. The level F _L4 is multiplied to obtain a feedback output F _O4 (S2007). The modulation output F _O4 is a modulation input to the next operator 3 (OP3).

Next, processing of operator 3 (OP3) is entered. This processing is almost the same as that of the operator 4 described above, except that there is no modulation input by the feedback output. First, the pitch data P ₃ is added to the current address A ₃ of the operation ₃ (S2008), and a transformation sign transformation is performed on the obtained value to obtain a carrier signal as O ₃ (S2009). Then obtained by, the O ₃ is obtained by adding the modulated output M _O4 above to become a new O ₃ (S2010), the value O ₃ is a triangle wave conversion and (S2011), In addition, the envelope value E is ₃ is multiplied by the output O ₃ (S2010) In addition, modulation. Level M _L3 is obtained in this (S2013). This modulation output _MO3 is a modulation input to the next operator 2 (OP).

Next, although the processing of the operator 2 (OP2) is executed, this processing is the same as in the case of the operator 3 described above because only the modulation inputs are different, and the description is omitted.

Finally, the process of operator 1 (OP1) is entered, but this process is the same as before until step S2024. The sound waveform output O ₁ obtained in S2024 is accumulated in the buffer B (see FIG. 12) as a carrier (S2025).

As mentioned above, although the embodiment of the normal sound source processing by the modulation method has been described, this processing is for one sounding channel as described above, and in fact, for each CPU of the M CPU 101 and the S CPU 102, Eight phonetic channels are processed (see also fourth (b)). If a modulation scheme is assigned to any sounding channel, the sound source processing by the above-described shifting scheme is moved.

[Examples of Modulation Modification (Part 1)]

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

The basic concept is shown in the flowchart of FIG.

In the figure, the operator, 1, 2, 3, and 4 processing only use variable names and the program structure is the same.

Here, the reason that each operator process cannot perform the operator process without determining the modulation input is because the modulation input to each operator process differs according to the algorithm as shown in FIG. In other words, it is necessary to determine which output of the algorithm processing is the modulation input, or whether the output of the algorithm processing is fed back as the modulation input rather than the algorithm processing. In the operation flow of FIG. 21, such connection relations are collectively performed as the algorithm processing (S2105), and the connection relations obtained thereby determine the modulation input at each operator processing (S2102-S2104) at the next interrupt timing. It works. In addition, an initial value is given as an input to each operator process at the start of pronunciation (i.e., the earliest).

In this way, if the operator processing separates the algorithm processing, in any algorithm, the program of the operator processing is the same, and the algorithm processing only needs to be changed. Therefore, it is possible to significantly reduce the program capacity of the entire sound source processing by the modulation method.

Next, an example of improvement of the FM modulation method based on the above-described basic concept will be described. The operation flowchart by the FM modulation method of the operator processing of FIG. 21 is shown in FIG. 22 (a) as an example of operator 1 processing, and the calculation algorithm per operator is shown in the same figure (b).

The other operators 2-4 processes are very similar just by different subscript numbers. Further, each variable in the flow is stored in a pronunciation channel somewhere in FIG. 9 on the RAMs 206 and 306 of the M CPU 101 or the S CPU 102.

First, the pitch data P ₁ is added to the address A ₁ corresponding to the phase angle to become a new address A ₁ (S2201). Next, the modulation input M ₁₁ is added to this address A ₁ , and an address A _M1 is obtained (S2202). The modulation input M ₁₁ is determined by the algorithm processing S2105 (21st) at the last interrupt timing, and by the algorithm, the operator's own feedback output F _O1 or the axis is output M of another operator, for example, operator 2. Maybe it was _O2 . Next, the value of the sine wave corresponding to an address (phase) A _M1 is read from the external memory 116 (FIG. 1), is the output O ₁ is obtained (S2203). After that, the value multiplied by the envelope data E ₁ becomes the output O ₁ of the operator 1 (S2204). This output O ₁ is multiplied by the gain.level F _L1 to obtain the gain output F _O1 (S2205), and the modulation and level M _L1 is multiplied by the output O ₁ to obtain the modulation M _O1 ( S2206).

Next, an example of improvement of the TM modulation method resulting from the basic concept mentioned above is demonstrated. The operation flowchart by the TM modulation method of the operator processing of FIG. 21 is shown in FIG. 23 (a) as an example of operator 1 processing, and the calculation algorithm per operator is shown in the same figure (b).

The other operators 2-4 processing are just the same, with different subscript numbers for the variables. Further, each variable in the flow is stored in a pronunciation channel somewhere in FIG. 9 on the RAMs 206 and 306 of the M CPU 101 or the S CPU 102.

First, pitch data P ₁ is added to the current address A ₁ (S2301). Next, the modified sine wave corresponding to the above-described address A ₁ (phase) is read from the external memory 116 (FIG. 1) by the modified sine transform fc, and a carrier signal is generated as O ₁ (S2302). The modulation input M ₁₁ is added as a modulation signal to the output O ₁ , and a new address O ₁ is obtained (S2303). Next, the value of the triangular wave corresponding to this address O ₁ (phase) is read from the external memory 116 (S2304), and the envelope value E ₁ is multiplied by this to become the output O ₁ (S2305). Thereafter, the feedback.level F _L1 is multiplied by this output O ₁ to obtain the feedback output F _O1 (S2306), and the modulation.level M _L1 is multiplied by the output O ₁ to obtain a modulation output M _O1 ( S2307).

Next, a specific example of the algorithm processing S2105 in FIG. 21 for determining the modulation input in the operator of both the FM system and the TM system described above will be described in the operation flowchart of FIG. The flow in the figure is common to the FM system and the TM system, and is an example in which 1-4 of the algorithm shown in FIG. 18 is switched. In this case, the selection branch of Algorithms 1-4 is selected based on an instruction not specifically shown by the player (S2400).

First, Algorithm 1 is a four-operator (abbreviated as OP) serial type as in the eighteenth (a), and only OP4 has a pitback input. That is, the pitback output F _O4 of the OP4 becomes the modulation input M ₁₄ of the OP4 (S2401), the modulation M _O4 of the OP4 becomes the modulation input M ₁₃ of the OP3 (S2402), and the modulation output M _O3 of the OP3 becomes the modulation of the OP2. and the input M12 (S2403), the modulation output of the OP M _O2 is added to and as OP1 modulation type M ₁₁ in (S2404), the buffer B outputs O ₁ of OP is a carrier output (see Fig. 12) (S2405) Is called an algorithm.

Algorithm 2 shows that OP2 and OP4 have a feedback input as shown in Fig. 18 (b), that is, feedback output F _O4 of OP4 becomes modulation input M ₁₄ of OP4 (S2406), and modulation output M _O4 of OP4 modulates OP3. Input M ₁₃ (S2407), feedback output F _O2 of OP2 becomes modulation input M ₁₂ of OP2 (S2408), modulation outputs M _O2 and M _O3 of OP2 and OP3 become modulation input M ₁₁ of OP1 (S2409) The output O ₁ of OP1 is added to the buffer B as a carrier output (S24010).

Algorithm 3 has OP2 and OP4 and feedback inputs, and two modulation series are configured in parallel. That is, the feedback output F _O4 of the OP4 becomes the modulation input M ₁₄ of the OP4 (S2411), the modulation output M _O4 of the OP4 becomes the modulation input M ₁₃ of the OP3 (S2412), and the feedback output F _O2 of the OP2 becomes the modulation of the OP2. Input M ₁₂ (S2413), modulation output M _O2 of OP2 becomes modulation input M ₁₁ of OP1 (S2414), and each output O ₁ and O ₃ of OP1 and OP3 is added to buffer B as a carrier output (S2415). ), Which is an algorithm.

Algorithm 4 is a four-operator parallel, with all operators having feedback inputs. That is, the feedback output F _O4 of the OP4 becomes the modulation input M ₁₄ of the OP4 (S2416), the feedback output F _O3 of the OP3 becomes the modulation input M ₁₃ of the OP3 (S2417), and the feedback output F _O2 of the OP2 becomes the modulation of the OP2. Input M ₁₂ (S2418), OP1's feedback output F _O1 becomes OP1's modulation input M ₁₁ (S2419), and all operators' outputs O ₁ , O ₂ , O _3, and O ₄ are added to buffer B ( S2420).

The sound source processing for one channel is finished by the operator processing and the algorithm processing described above, and if the algorithm is not changed, the pronunciation (sound source processing) is continued in this state.

[Example of Improvement of Modulation Method (Part 2)]

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

In the various modulation methods described so far, processing takes longer as the complex algorithm is organized and the number of pronunciation channels (polyphonic number) is larger.

Therefore, in the second improved example described below, the first improved example of FIG. 21, which separates the operator process and the algorithm process, is further developed, and only the operator process is performed at an interrupt timing, and the algorithm process is performed at the next interrupt timing. The operator process and the algorithm process are alternately performed as if by only doing this. As a result, the processing load per one interrupt timing can be significantly reduced. As a result, one sample data is output at two interrupts.

This operation will be described using the operation flowchart of FIG.

First, since the operator process and the algorithm process are alternately performed, it is determined whether or not S is zero using the variable S (S2501). This variable S is formed for each pronunciation channel and is stored in each pronunciation channel region of FIG. 9 on the RAMs 206 and 306 of the M CPI 101 or the S CPU 102.

At any interrupt timing, when S is 0, it enters the root of the operator process and sets the value 1 to the variable S (S2502). Next, the operator 1-4 processing is executed (S2503-S2506). This processing is the same as in FIG. 22, 23, and the like.

Next, the root of the operator process is extracted, and an output process for setting the value of the buffer BF (in the FM method) or the buffer BT (in the TM method) is executed in the buffer B (S2510). A buffer BF or BT is formed for each pronunciation channel, and is stored for each pronunciation channel region of FIG. 9 on the RAMs 206 and 306 of the M CPU 101 or the S CPU 102. The waveform output values after the algorithm processing are stored in the buffer BF or BT. However, since the algorithm processing is not executed at this interrupt timing and the contents of the buffer BF or BT are not changed, the waveform output values same as the previous interrupt timing are output. .

The above processing ends the sound source processing for one sound channel at this interrupt timing. In this case, each data obtained by the operator 1-4 processing is held in each sounding channel region of FIG. 9 on the RAMs 206 and 306 of the M CPU 101 or the S CPU 102 until the next interrupt timing.

If the next interrupt is applied, since the variable S is 1 at the previous interrupt timing, the flow advances to step S2507, enters the root of the algorithm processing, and sets the value 0 to the variable S. FIG.

Then, the algorithm processing is executed (S2508).

In this processing, each data processed in the operator 1-4 processing at the previous interrupt timing and held in each pronunciation channel region in FIG. 9 is used, and a process for determining the modulation input for the next operator processing is performed. In this process, the contents of the buffer BF or BT are improved, and the waveform output value is obtained at the interrupt timing. An example of the algorithm processing is shown in the operation flowchart of FIG. In this flow, in the step of numbering the same as in the case of FIG. 24, the process is performed as in the case of FIG. What is different from the case of FIG. 24 is the output portion of S2601-S2604. Here, in the case of Algorithm 1 and Algorithm 2, the contents of the output O ₁ of the operator 1 processing are held in the buffer BF or BT as they are (S2601, S2602). In the case of Algorithm 3, the output O ₃ are added to the output value O ₁ is maintained in the buffer BF or BT (S2603). In the case of Algorithm 4 is the output O _2, O _3, O ₄ added to the output value O ₁ is maintained in the BF or BT (S2604).

As described above, since the operator process and the algorithm process are alternately performed every other interrupt timing, the processing load of the sound source processing program per interrupt time can be significantly reduced. In this case, since the interrupt period does not need to be lengthened, the time for interrupting the main operation flowchart of FIG. 4 (a) by interruption does not increase, and the processing load can be reduced without affecting the program operation. . Therefore, for example, the interval between the key-key pickings performed by FIG. 4 (a) does not become long and does not affect the response performance as an electronic musical instrument.

In the above, the operation | movement which generate | occur | produces the sound tone data for every pronunciation channel by the sound source process of the software which originated in various timbre systems was demonstrated.

[Function Key Processing]

Next, the specific operation of the function key processing S403 of the main operation flowchart of FIG. 4 (a) in the case of playing the actual electronic musical instrument will be described.

In the sound source processing performed for each pronunciation channel described above, for example, the function key shown in FIG. 8 (a) connected to the operation panel of the electronic musical instrument through the input port 210 (see FIG. 2) of the M CPU 101. 801 denotes each of the pronunciation channel regions (see FIG. 9) on the RAM 206 or 306 (see FIG. 2 and FIG. 3) of the M CPU 101 or the S CPU 102. Data formats (see FIG. 11) are set in various sound source systems.

FIG. 27 is a diagram showing an arrangement example of a part of the function keys 801 in FIG. 8 (a). In the figure, a part of the function key 801 is realized as a tone designation switch, and when a switch such as "piano", "guitar", .... "lyre" of group A is pressed, the tone of each musical instrument sound is pressed. Is selected, and the guide lamp is lit. Then, with the DPCM / TM method selection switch 2701, it is selected which sound source method of the DPCM method and the TM method produces the tone of these musical instrument sounds.

On the other hand, each tone is assigned to the FM method when the "Tub" switch of Group B is pressed, by the PCM / TM method when "Bass" is pressed, and by the PCM method when "Trumpet" is pressed. Then, the musical sound due to their sound source system is generated.

Assignment of the sound source system to each pronunciation channel region shown in FIG. 9 on RAM 206 or 306 when the " piano " and " bass " are pressed in FIGS. 28 (a) and (b) An example is shown. In the case of " piano ", the DPCM method is assigned to all the pronunciation channels of the eight sound polyphonic of the M CPU 101 and the S CPU 102, as shown in the diagram (a) of FIG. In this case, as shown in (b), the PCM method is assigned to the pronunciation channel of the odd number and the TM method is assigned to the pronunciation channel of the even number, respectively. As a result, the sound wave waveform generated in the two sound channel by the PCM method and the TM method is mixed, so that one sound wave waveform is obtained. In this case, 4 sound polyphonic and 8 CPU polyphonic for each CPU are added.

FIG. 29 is a part of the operation flowchart of the function key processing of S403 in the main operation flowchart of FIG. 4 (a), and is an operation flowchart of the processing for the tone tone designation switch group of FIG.

First, it is determined whether or not the DPCM / TM switch 2701 is operated by the player (S2901). If the determination is YES, it is determined whether the variable M is zero or not (S2902). The modulation M is secured on the RAM 206 (FIG. 2) of the M CPU 101, and a value 0 is obtained for the DPCM method and a value 1 for the TM method. If the value of the variable M is 0 in S2902 and the determination is YES, the value 1 is set in the variable M (S2903). This means that in the state where the DPCM system was selected, the DPCM / TM switch 2701 was pressed and changed to the state in which the TM system was selected.

If the value of the variable M is determined to be 1 in S2902, 0 is set in the variable M (S2904). This means that in the state where the TM system was selected, the DPCM / TM switch 2701 was pressed and changed to the state where the DPCM system was selected.

Next, it is determined whether or not the tone of the group A in Fig. 27 is specified (S2905). Since the DPCM / TM switch 2701 is valid only for the tones of group A, the operation corresponding to the DPCM / TM switches 2701 of S2904-S2908 is only performed when the tones of the group A are designated and the judgment of S2905 is YES. Is executed.

In S2906, it is determined whether the variable M is the value 0 (S2906).

If the determination of S2906 is YES, since the DPCM method is selected by the DPCM / TM switch 2701, each of the pronunciation channel regions on the RAM 206 and 306 (see Figs. Data is set in the DPCM format of FIG. In other words, the sound source method No. indicating the DPCM method is set in the head region G (see the column of DPCM in FIG. 11) of each pronunciation channel region. Subsequently, various parameters corresponding to the timbre currently specified in the second and subsequent regions of the pronunciation channel region are set, respectively (S2907).

If the determination of S2906 is NO, since the TM method is selected by the DPCM / TM switch 2701, data is set in the TM format shown in Fig. 11 in each pronunciation channel region. That is, first, a sound source method No. indicating the TM method is set in the head area G of each pronunciation channel region. Subsequently, various parameters corresponding to the tone currently specified in the second and subsequent regions of the pronunciation channel region are set, respectively (S2908).

The above is the case where the DPCM / TM switch 2701 of FIG. 27 is operated, but when the switch is not operated and the judgment of step S2901 is NO, or the tone of the group A is not specified, the judgment of step S2905 is performed. If NO, the process of step S2909 or less is executed.

First, in step S2909, it is determined whether or not there is a change in the tone switch of Fig. 27 (S2909).

If the determination is NO, it is not necessary to perform the processing for the tone switch, and thus the function key processing (S4 (a) also ends S403).

If there is a change in the tone switch and the determination in S2909 is YES, it is then determined whether or not the tone of the group B is designated (S2910).

If the tone of group B is designated and the determination of S2910 is YES, the data is output in the sound source method data format corresponding to the tone assigned to each pronunciation channel area on RAM 206 and 306 (see FIGS. 2 and 3). Is set. Then, a sound source method No. indicating each sound source method is set in the head region G (see FIG. 11) of each pronunciation channel region. Further, various parameters corresponding to the tone tone currently designated in the second and subsequent regions of the pronunciation channel region are set, respectively (S2911). For example, when the base switch shown in Fig. 27 is selected, data corresponding to the PCM method is set in each phonetic channel region of the odd number, and data corresponding to the TM system is set in each phonetic channel region of the even number.

If the tone switch of the group A is designated and the determination of S2910 is NO, it is determined whether or not the variable M is 1 (S2912). If the determination of S2912 in which the TM method is currently selected is YES, similarly to the case of step S2908 described above, data is set in the TM format of FIG. 11 in each sounding channel region (S2913).

If the determination of S2912 in which the DPCM method is selected is NO, similarly to the case of step S2907 described above, data is set in each DPCM format in the DPCM format of FIG. 11 (S2914).

[First Embodiment of Key Key Processing at the Time of Drying]

Next, the specific operation of the key-key processing (S405) of the main operation flowchart in FIG. 4 (a) when playing the actual electronic musical instrument will be described.

First, a description will be given of a first embodiment of keyboard key processing at the time of pressing.

In the first embodiment of the keypress key processing at the time of dryness, the eighth connected via the input port 210 (see FIG. 2) of the M CPU 101 when the tone of the group A in FIG. 27 is designated. RAM 206 of the M CPU 101 or the S CPU 102 depending on the position of the keyboard in the case where the key 802 shown in FIG. The sound source system set in each pronunciation channel region (see FIG. 9) on 306 (see FIGS. 2 and 3) is automatically switched. In this case, with the key code numbers 31 and 32 of the key 801 shown in FIG. 8 (b) as a boundary, in the case of the low range of the key code 31 or less of the keyed key, the DPCM method is used as a key. The TM scheme is assigned when the chord is 32 or more highs. In addition, when the tone of the group B of FIG. 27 is designated, no special key key processing is executed.

FIG. 30 is a part of the operation flowchart of the keyboard key processing of S405 in the main operation flowchart of FIG. 4 (a), and the key key 801 of FIG. 8 (a) or FIG. This is the flow of the first embodiment of the process.

First, it is determined whether or not the tone of the group A in FIG. 27 is currently designated (S3001).

If this determination is NO and the tone color of group B is currently specified, no special processing of Fig. 20 is performed.

If the determination of S3001 is YES and the tone of group A is currently designated, the keyboard of the keyboard determined to be " presse " in the keyboard key extraction processing of S _{404 in} the main operation flowchart of FIG. It is determined whether or not the number of times is less than (S3002).

If the low frequency range below 31 is pushed in and the determination of S3002 is YES, it is determined whether or not the variable M is the value 1 (S3003). The variable M is set in the operation flowchart of FIG. 29 which is a part of the function key processing of S403 in the main operation flowchart of FIG. 4 (a), and the value of the variable M as described above is the value I in the DPCM method. In case of TM, value 1 is obtained.

If the determination of S3003 is YES (M = 1), and the sound source method of TM method is currently designated, RAM 206 or 306 (FIG. 2) in order to change to DPCM method which is a low sound source method. And data are set in the DPCM format of FIG. 11 in the pronunciation channel region to which the pressed key on FIG. 3 is assigned. In other words, the sound source method No. indicating the DPCM method is set in the head region G (see the column of DPCM in FIG. 11) of each pronunciation channel region. Subsequently, various parameters corresponding to the tone tone currently specified in the second and subsequent regions of each pronunciation channel region are set (S3004). Thereafter, the value 1 is set in the flag C (S3005). The flag C is a variable secured to each pronunciation channel area on the RAM 206 (see FIG. 2) of the M CPU 101 (see FIG. 11), but this flag C is a process of this case described later in FIG. Used for

On the other hand, in S3002, when 31 or more high-range ranges are pushed in and the determination is NO, it is again determined whether or not the variable M has a value of 1 (S3006).

If the determination of S3006 is NO (M = 0) and the sound source method of the DPCM method is currently designated, the keyed key on the RAM 206 or 306 is changed to the TM method of the high sound source method. In the pronunciation channel region to be assigned, data is set in the TM format of FIG. In other words, the sound source method No. indicating the TM method is set in the head region G (see the column of TM in FIG. 11) of each pronunciation channel region. Subsequently, various parameters corresponding to the timbre currently specified in the second and subsequent regions of the pronunciation channel region are set, respectively (S3007). Thereafter, the value 2 is set in the flag C (S3005).

In the above-described processing, when the determination in step S3003 is NO and the determination in step S3006 is YES, since the sound source system is originally desired, no special processing is performed.

[Second embodiment of the dry key processing at the time of dryness]

Next, a description will be given of a second embodiment of the dry key processing at the time of dryness.

In the second embodiment of the keypress key processing at the time of dryness, when the tone of the group A in FIG. 27 is designated, the eighth connected via the input port 210 of the M CPU 101 (see FIG. 2). The RAM 206 of the M CPU 101 or the S CPU 102 by the dry speed, i.e., Velocity, in the case where the dry key 802 shown in Figs. (a) or 8 (b) is pushed. Alternatively, the sound source system set in each pronunciation channel region (see FIG. 9) on 306 (see FIGS. 2 and 3) is automatically switched. In this case, the DPCM method is used in the case of fast dry operation where the value of the speed of the keyed keyboard is 64 or more, with 64 being the limit of 64 of the maximum 127 of the Musical Instrument Didital Interface (MIDI) standard. In the case of a slow dry operation where the velocity value is less than 64, the TM method is assigned. In addition, when the tone of the group B of FIG. 27 is designated, no special key key processing is executed.

In the main operation flowchart of FIG. 31 (a) of FIG. 31, it is a part of the operation flowchart of the keyboard key processing of S405, and the key key 801 of FIG. 8 (a) or FIG. This is the flow of the second embodiment of the process.

First, it is determined whether or not the tone of the group A in FIG. 27 is currently designated (S3101).

If this determination is NO and the tone color of group B is currently specified, no special processing of FIG. 30 is performed.

If the judgment at S3101 is YES and the tone of group A is currently designated, the speed of the key determined as "press" is 64 in the key-key extraction process of S404 in the main operation flowchart of FIG. 4 (a). It is determined whether or not it is abnormal (S3102). In addition, 64 of speed is equivalent to mp (mezzo piano) of MIDI standard.

If the determination of S3102 whose value of the speed is 64 or more is YES, it is determined whether or not the variable M is the value 1 (S3103). The variable M is set in the operation flowchart of FIG. 29 which is part of the functional processing of S403 in the main operation flowchart of FIG. 4 (a). As described above, the value of the variable M is 0, TM in the DPCM method. In the case of a method, the value 1 is obtained.

If the determination of S3103 is YES (M = 1), and the sound source method of the TM method is currently designated, S3004 in the case of the first embodiment of FIG. 30 in order to change to the DPCM method which is the sound source method at the time of a fast pressing operation. As in the processing of the data, data is set in the DPCM format of FIG. 11 in the pronunciation channel region to which the confiscated key on the RAM 206 or 306 (see FIGS. 2 and 3) is assigned (S3104). The value 1 is set to C (S3104).

On the other hand, in S3102, when the value of the speed is smaller than 64 and the determination is NO, it is again determined whether or not the variable M is 1 (S3106).

If the determination of S3106 is NO (M = 0) and the sound source method of the DPCM method is currently designated, S3007 in the case of the first embodiment of FIG. 30 to change to the TM method, which is the sound source method at the time of a slow pressing operation. In the same manner as in the case of the processing of, the data is set in the TM format of FIG. 11 in the pronunciation channel region to which the pressed key on the RAM 206 or 306 is assigned (S3107), and the value 2 is set in the flag C. (S3108).

In the above-described processing, when the determination in step S3103 is NO and the determination in step S3106 is YES, since the sound source system is originally desired, no special processing is performed.

[Example of this key key processing]

Next, the Example of keyboard key processing at this time is demonstrated.

According to the first or second embodiment of the key key processing at the time of the pressing operation, the sound source system can be automatically changed by the keyboard area (tone range) or speed, but in this case, their change settings are returned to their original state. There is a need. Realizing it is an embodiment of key key processing of the present invention described below.

In the main operation flowchart of FIG. 32 (a) of FIG. 32, it is a part of the operation flowchart of the keyboard key processing of S405, and this time of the keyboard key 801 of FIG. 8 (a) or FIG. It is a flow of processing.

First, in the main operation flowchart of FIG. 4 (a), the RAM 206 or 306 to which the key determined as "this" in the key-key extraction process of S404 is assigned (see FIGS. 2 and 3). The value of the flag C set in the pronunciation channel region on the? Now, the flag C is set in the processing of S3005 and S3008 of FIG. 30 or S3105 or S3108 of FIG. 31, the initial value at the time of pressing is 0, and the value 1 is changed when the sound source method is changed from TM method to DPCM method at the time of pressing The value 2 is set when the sound source method is changed from the DPCM method to the TM method. Therefore, when no change is made at the time of pressing, the initial value is 0.

In the processing at this time in FIG. 32, when the value of the flag C is 0 by the determination at step S3201, the sound source method is not changed by the keyboard area or the speed, so that no special processing is performed and the normal processing is performed. Is done.

If it is determined by S3201 that the value of the flag C is 1, the sound source method is changed from the TM method to the DPCM method at the time of pressing. Thereby, in order to return to the TM system, data in the TM format of FIG. 11 is transferred to the pronunciation channel region to which the compressed key on the RAM 206 or 306 (see FIG. 2 or FIG. 3) is assigned. Is set. In other words, the sound source method No. indicating the TM method is set in the head region G (see the column of TM in FIG. 11) of each pronunciation channel region. Subsequently, various parameters corresponding to the timbre currently specified in the second and subsequent regions of each pronunciation channel region are set (S3202).

On the other hand, when it is determined by S3201 that the value of the flag C is 2, it is changed from the sound source system DPCM to the TM system at the time of pressing. Therein, in order to return to the program method, data is set in the DPCM format of FIG. 11 in the pronunciation channel region to which the confiscated key on the RAM 206 or 306 is assigned. In other words, the sound source method No. indicating the DPCM method is set in the head region G (see the column of DPCM in FIG. 11) of each pronunciation channel region. Subsequently, various parameters corresponding to the timbre currently specified in the second and subsequent regions of each pronunciation channel region are set (S3203).

After the above operation, the value of the flag C is returned to zero, the processing of FIG. 32 is terminated, and then, in particular, the ordinary transfer processing (not shown) is executed.

[Status of Other Embodiments]

In the series of embodiments of the present invention described above, as shown in FIG. 1, two CPUs, M CPU 101 and S CPU 102, share different pronunciation channels and process them. May be one or three or more.

The control ROMs M201 and 301 and the external memory 116 of FIGS. 2 and 3, and the external memory 116 are constituted by a ROM card and the like, and various sound source methods can be provided to the user by the ROM card.

In addition, it is possible to connect various operation units to the input port 210 of the M CPU 101 of FIG. 2 in addition to the musical instrument operation unit as shown in FIG. 8, thereby realizing various types of electronic musical instruments. Moreover, it is also easy to implement | achieve as a sound source module which inputs performance information from another electronic musical instrument, and performs only a sound source process.

On the other hand, the form in which the sound source method is assigned to each pronunciation channel by the function key 801 or the keyboard key 802 shown in FIG. 8 can be considered in various forms including tone tone, sound range, and speed.

As the modulation method, various ones can be applied in addition to the FM and TM methods.

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

As described above, according to the present invention, a dedicated sound source circuit is not required at all, and a general processor configuration can be achieved. In this way, the circuit size of the entire acoustic wave generator can be greatly reduced, and even in the case of LSI, it is the same as the manufacturing technology of a normal microprocessor, and the ratio of the product to the raw material of the chip also improves the manufacturing cost. It is possible to greatly reduce. In addition, since the sound signal output means cannot be constituted by a simple chip circuit, there is little increase in manufacturing cost by adding this portion.

In addition, for example, when changing the modulation method to the phase modulation method and the frequency modulation, or to change the number of polyonics, it is possible to cope by simply changing the sound source processing program to be stored in the program storage means. It is possible to greatly reduce the development cost of the acoustic wave generator, and to provide a new sound source method to the user, for example, by a ROM card.

In this case, a program for executing the sound source processing program at predetermined time intervals for the data architecture and the performance information processing program which performs the link of data between the performance information processing program and the sound source processing program through the achromatic data on the data storage means. By realizing the architecture, a process for obtaining synchronization between both processors is unnecessary, and the program can be greatly simplified. As a result, sound source processing that is complicated in processing as in the modulation method can be executed with sufficient margin.

In addition, since a change in processing time due to selection of a sound generation algorithm in the type of modulation method or modulation method can be absorbed by the sound signal output means, a complicated time for outputting a sound signal to a D / A converter or the like can be obtained. It also has the effect of not needing a dimming control program.

In addition, in the present invention, as an architecture of the sound source processing program, after the at least one operator process, which is a modulation processing unit, is executed in a batch or before its execution, a processing structure in which the algorithm processing, which is each input / output process between each operator process, is to be executed in a batch. Therefore, even when the sound source processing is performed by selecting from among a plurality of algorithms in the modulation method, only a plurality of types of algorithm processing can be easily performed, and it can be switched as necessary, and the sound source processing program can be made very compact. In terms of program capacity, it contributes to miniaturization and low cost of the acoustic wave generator.

Claims (23)

1. Memory means 201 for storing a sound source processing program relating to a modulation method, and sound signal generating means 101 for generating a sound signal as a modulation process by executing the sound source processing program stored in this storage means. And sound signal generating means for outputting the sound signal generated by the sound signal generating means at a constant output time interval. 2. The digital sound signal according to claim 1, wherein the sound signal output means (107, 108) comprises a timing signal generating means (203) for generating a timing signal of each predetermined sampling period and the digital sound signal from the sound signal generating means (101). First latch means (601) for latching the digital musical sound generated by the sound signal generating means at the output timing of the first latch means (601), and when the timing signal is generated from the timing signal generating means (203). And second latch means (602) for outputting the digital sound signal by latching an output signal of the sound wave waveform. The sound signal generating means according to claim 1, characterized in that the sound signal generating means (101) executes a process of outputting a mixed signal obtained by mixing a modulation signal to a carrier signal as an input signal and outputting a modulated sound signal as an output signal. Waveform Generator. The method of claim 1, wherein the sound signal generating means 101 performs a process of outputting a mixed signal obtained by mixing a modulation signal to a carrier signal as an input signal, and outputting the modulated sound signal as an output signal. The functional relationship between the signal is neither a sine function nor a cosine function, and the carrier signal is a signal that causes the output signal to be a sine wave or cosine wave of a single frequency when the modulated signal is not mixed. Sound wave waveform generator, characterized in that. The method of claim 1, wherein the sound signal generating means 101 performs a process of outputting a mixed signal obtained by mixing a modulation signal to a carrier signal as an input signal, and outputting the modulated sound signal as an output signal. A sound wave generator, characterized in that the functional relationship with the signal is a sine function and the carrier signal is a sine wave. Program storage means (201) for storing a performance information processing program for processing performance information and a sound source processing program by a modulation method for obtaining a sound signal, an address control means (205) for controlling an address of the program storage means; Data storage means 206, arithmetic processing means 208, 209, the address control means 205, and the data storage means 206 for storing sound generation data necessary for generating a sound tone signal by the modulation method. And means for executing the performance information processing program or the sound source processing program stored in the program storage means 201 while controlling the arithmetic processing means 208, 209. Control the sound generation data on the data storage means to be executed, and transfer the control to the sound source processing program at a predetermined time interval. Means for executing the performance information processing program again after the completion of the sound source processing program, and at the time of execution of the sound source processing program, executing a program for generating a sound signal by the modulation method on the basis of the sound generation data on the data storage means. Music sound signal output means 107, 108 for holding the sound signal obtained by executing the means 101, the program execution means 101, and the sound source processing program, and outputting the held sound signal at regular output time intervals. Sound wave waveform generator characterized in that it has a). 8. The program execution means (101) according to claim 6, wherein said program execution means (101) includes interrupt control means (203) for generating an interrupt signal at said predetermined time interval, and said program execution means (101) is executing said performance information processing program. The interrupt control means 203 stops the performance information processing program at the timing at which the interrupt signal is generated, transfers control to the sound source processing program, executes it, and releases the interrupt after the termination of the performance information processing program. A sound wave generator, characterized by resuming execution. 7. The digital sound source signal according to claim 6, wherein the rooting processing output means (107, 108) comprises a timing signal generating means (203) for generating timing signals of each predetermined sampling period, and the program execution means (101). First latch means 601 for latching the digital sound signal generated by the program executing means 101 at an output timing of the first latch means, and when the timing signal is generated by the timing signal generating means 203, the first latch. And a second latch means (602) for outputting a digital sound signal by latching an output signal of the means (601). The program execution means (101) according to claim 6, wherein said program execution means (101) outputs a mixed signal obtained by mixing a modulation signal to a carrier signal as an input signal, and outputs a modulated sound signal as an output signal by execution of a sound source processing program. Sound wave waveform generator characterized in that. 7. The processing according to claim 6, wherein said program executing means (101) outputs a mixed signal obtained by mixing a modulation signal to a carrier signal as an input signal and an output of a modulated sound signal as an output signal by execution of a sound source processing program. The function relationship between the input signal and the output signal is neither a sine function nor a cosine function, and the carrier signal is a sinusoidal wave having a single frequency when the output signal is not mixed with the modulated signal. Or a sound wave generator that is a signal to be a cosine wave. 7. The process according to claim 6, wherein the program generating means (101) outputs a mixed signal obtained by mixing a modulation signal to a carrier signal as an input signal and an output modulated sound signal as an output signal by execution of a sound source processing program. Wherein the functional relationship between the input signal and the output signal is a sine function, and the carrier signal is a sine wave. Storage means 201 for storing a sound source processing program relating to a transfer method having an operator processing program for executing at least one operator process and an algorithm processing program for executing an algorithm process for determining an input-output relationship between the respective operator processes; Sound signal signal generating means 101 for generating a sound signal by independently carrying out an independent collective processing of the operator processing based on the operator processing program and the algorithm processing based on the algorithm processing program, and by the sound signal generating means Sound wave signal generating device characterized in that it has a sound signal output means (107, 108) for outputting the sound signal at regular intervals. 13. The digital sound signal according to claim 12, wherein the sound signal output means (107, 108) comprises a timing signal generating means (203) for generating a timing signal of each predetermined sampling period and the digital sound signal from the sound signal generating means (101). First latch means 601 for latching the digital sound signal generated by the sound signal generating means 101 at the output timing of the signal, and the first signal when the timing signal is generated from the timing signal generating means 203. And a second latch means (602) for outputting a digital sound signal by latching an output signal of the latch means (601). The sound wave generator according to claim 12, wherein said sound signal generating means (101) outputs a modulated sound signal as an output signal as a mixed signal obtained by mixing a modulation signal with a carrier signal. The method of claim 12, wherein the sound signal generating means 101 performs a process of outputting a modulated sound signal as an output signal by inputting a mixed signal obtained by mixing a modulation signal with a carrier signal as an input signal. The functional relationship between is not a sine function or a cosine function, and the carrier signal is a signal that causes the output signal to be a sine wave or cosine wave of a single frequency when the modulated signals are not mixed. Sound wave waveform generator. The method of claim 12, wherein the sound signal generating means 101 performs a process of outputting a modulated sound signal as an output signal by inputting a mixed signal obtained by mixing a modulation signal with a carrier signal as an input signal. And a functional relationship between is a sinusoidal function, and the carrier signal is a sinusoidal wave. In a performance information processing program for processing performance information and a sound source processing program using a modulation method for obtaining a sound signal, each operator processing after at least one operator, which is a modulation processing unit, is collectively executed or before execution thereof. Program storage means 201 for storing a sound source processing program having a processing structure in which algorithm processing, which is input / output processing between respective operator processes, is executed collectively after or before execution thereof; Address control means 205 for controlling an address, data storage means 206 for storing sound generation data necessary for generating a sound signal by the modulation method, arithmetic processing means 208, 209, and the address control The program 205 controls the data storage means 206 and the arithmetic processing means 208,209. Means for executing the performance information processing program or the sound source processing program stored in the gram memory means 201, and normally executes the performance information processing program to control the sound generation data on the data storage means 206; Means for transferring control to the sound source processing program at predetermined time intervals and executing it, and again executing the performance information processing program after the completion of the sound source processing program. Program execution means (101) for generating a sound signal in the modulation method based on the sound generation data on the data storage means (206) in the program execution means for generating a sound signal in the modulation method based on data; The sound sound image obtained by the program executing means 101 executing the sound source processing program The maintenance, and the maintenance of the musical tone signal output means for generating musical tone waveforms and wherein having a (107, 108) for outputting a sound signal with a constant output time interval. 18. The program execution means (101) according to claim 17, wherein said program execution means (101) includes interrupt control means (203) for generating an interrupt signal at said predetermined time interval, and said program execution means (101) is executing said performance information processing program. The performance control program stops the performance information processing program at the timing when the interrupt signal is generated from the interrupt control means 203, transfers control to the sound source processing program, executes it, and releases the interrupt after the execution of the performance information processing program. Sound wave waveform generator, characterized in that through. 18. A plurality of algorithms according to claim 17, wherein the sound source processing program according to the modulation method is each input / output process between the respective operator processes after at least one operator process which is a modulation processing unit is executed collectively or before its execution. A sound wave generator, characterized in that it has a processing structure which is collectively executed during processing. 18. The apparatus according to claim 17, wherein the sound source processing output means (107, 108) comprises a timing signal generating means (203) for generating a timing signal of each predetermined sampling period, and digital from the program executing means (101). First latch means (601) for latching the digital music signal generated by the program executing means at the output timing of the sound source processing, and the first latch means (1) when the timing signal is generated from the timing signal generating means (203). And a second latch means (602) for outputting a digital sound signal by latching an output signal of the signal (601). The program execution means (101) according to claim 17, wherein the program executing means (101) outputs a mixed signal obtained by mixing a modulation signal to a carrier signal as an input signal and an output modulated sound signal as an output signal by execution of a sound source processing program. Sound wave waveform generator. The process according to claim 17, wherein said program execution means (101) outputs a mixed signal obtained by mixing a modulation signal to a carrier signal as an input signal, and outputs a modulated sound signal as an output signal by execution of a sound source processing program. Wherein the functional relationship between the input signal and the output signal is neither a sine function nor a cosine function, and the output signal is single when the carrier signal is not mixed with the modulation signal. A sound wave generator, characterized in that the signal is a sinusoidal or cosine wave of frequency. The program generating means (101) according to claim 17, wherein the program generating means (101) performs a process of outputting a mixed signal obtained by mixing a modulation signal to a carrier signal as an input signal and outputting a modulated sound signal as an output signal by executing a sound source processing program. Wherein the functional relationship between the input signal and the output signal is a sine function, and the carrier signal is a sine wave.