JPH0460697A - Musical sound waveform generator - Google Patents

Abstract

PURPOSE:To miniaturize a circuit by performing a time division processing based on musical sound production data with respect to each sound production channel to generate a musical sound signal at the time of executing a sound source processing program and mixing musical sound signals in each of plural output groups. CONSTITUTION:A program executing means 207 normally executes a playing information processing program to control corresponding musical sound production data on a data storage means 206 and performs the time division processing based on musical sound production data on the data storage means 206 with respect to each sound production channel at the time of executing a sound source processing program to generate a musical sound signal corresponding to the sound production channel. Each sound production channel is allowed to correspond to one of plural output groups, and musical sound signals generated in sound production channels included in each output group are mixed to generate a musical sound signal output of the output group. Consequently, a general-purpose processor constitution is realized without requiring a private sound source circuit. Thus, the circuit scale is miniaturized.

Description

[Detailed Description of the Invention] [Industrial Application Field] The present invention relates to a sound source processing method in a musical waveform generator, and more specifically, to a musical sound source processing method that can output musical sound signals separately for each output group corresponding to a stereo channel or the like. This invention relates to a waveform generator.

[Conventional technology]

With the development of digital signal processing technology and LSI processing technology, various electronic musical instruments with high performance have been realized.

Musical sound waveform generators for electronic musical instruments require a large amount of high-speed digital calculations, so conventionally they were constructed using a dedicated sound source circuit that realized a musical sound generation algorithm based on the required sound source method and an equalization architecture in hardware. has been done. With such a sound source circuit, a sound source method based on PCM modulation or a modulation method is realized.

The above-mentioned sound source circuits are large in scale regardless of the sound source type. When implemented as an LSI, the scale will be about twice that of a general-purpose data processing microprocessor, even excluding the memory portion that stores PCM waveform data or modulation waveform data.

The reason for this is that the tone generator circuit requires complex atlus control to access waveform data based on various performance information. Another reason is that registers and the like for temporarily holding intermediate data obtained in the process of sound source generation processing need to be placed at various locations in an architecture compatible with the sound source method. Furthermore, in order to realize a polyphonic configuration that can generate a plurality of musical tones in parallel, shift registers and the like for performing time-sharing hardware processing of the sound source are also required at various locations.

[Problem to be solved by the invention]

As mentioned above, conventional musical waveform generators are configured with dedicated sound source circuits that are compatible with the sound source method.
The problem is that the hardware scale increases, and if it is realized by LSI, the cost increases at the manufacturing stage in terms of yield during LSI chip manufacturing, and the musical waveform generator becomes larger. have.

Furthermore, when it is desired to change the sound source system, the number of polyphonics, etc., a significant change in the sound source circuit is forced, resulting in an increase in cost at the development stage.

When realizing a conventional musical sound waveform generator as an electronic musical instrument, it is necessary to generate data that can be processed by a sound source circuit from performance information corresponding to performance operations, or to generate performance information that can be used with other musical instruments. A control circuit composed of a microprocessor or the like is required for communication. In such a control circuit, in addition to a performance information processing program for processing performance information, a sound source control program corresponding to the sound source circuit is required for supplying data corresponding to the performance information to the sound source circuit. , Moreover, it is necessary to operate both programs in synchronization. Due to the complexity of such a program, there is a problem in that the development cost increases significantly.

On the other hand, in recent years, many high-performance microprocessors for general-purpose data processing have been realized. It is also conceivable to realize a device. However, there is no known technique for synchronizing and operating a performance information processing program for processing performance information and a sound source processing program for executing sound source processing based on the performance information. In particular, since the processing time in the sound source processing program changes depending on the sound source method, a complicated timing control program is required to output the generated musical tone data to the D/A converter. As described above, if sound source processing is simply performed using software, the processing program becomes extremely complicated, and sophisticated sound source processing cannot be performed in terms of processing speed and program capacity. In particular, in order to give the audience a sense of presence and to enable output to multiple lines during stage performances and studio recordings, the musical sound signals generated by each sound generation channel are arbitrarily mixed to produce stereo left and right channels. It is not possible to perform advanced sound source processing such as outputting musical sound signals to multiple output groups such as multiple lines or multiple lines.

The present invention makes it possible to arbitrarily mix the musical tone signals of each sound generation channel between the sound generation channels and output them to multiple output groups using a microprocessor's program control, without requiring a dedicated sound source circuit. The purpose is to

[Means for Solving the Problems] First, the present invention includes a program storage means such as a ROM that stores a performance information processing program for processing performance information and a sound source processing program for obtaining musical tone signals. The sound source method in this case is, for example, a PCM method, a phase modulation method, a frequency modulation method, or the like.

Next, it has address control means for controlling the address of the program storage means.

It also includes data storage means for storing musical tone generation data necessary for generating musical tone signals for each sound generation channel.

Furthermore, it has an arithmetic processing means including a multiplier that executes four arithmetic operations.

The program execution means executes the performance information processing program or the sound source processing program stored in the program storage means while controlling the address control means, data storage means, and arithmetic processing means described above. The same means is
Normally, the performance information processing program is executed to control the corresponding musical tone generation data on the data storage means, control is transferred to the sound source processing program at predetermined time intervals, and executed, and after the performance information processing program is finished, the performance information processing is resumed. Run the program. Furthermore, when executing the sound source processing program, the program execution means performs time-sharing processing for each sound generation channel based on the musical tone generation data on the data storage means to generate a musical tone signal corresponding to each sound generation channel. Then, each of the sound generation channels is made to correspond to one of a plurality of output groups, and for each of the output groups, the musical tone signals generated by the sound generation channels included in the sound generation channels are mixed together, and the musical tone signals of the respective output groups are mixed. Generate signal output. The plurality of output groups in this case correspond to, for example, two-channel or four-channel stereo output channels. In addition, simple line output may be supported. In this case, the program execution means includes, for example, interrupt control means for generating an interrupt signal at the predetermined time intervals. As a result, the program execution means interrupts the performance information processing program at the timing when an interrupt signal is generated from the interrupt control means while executing the performance information processing program, transfers control to the sound source processing program, executes it, and executes the performance information processing program. After completion, the interrupt is canceled and execution of the performance information processing program is resumed.

In addition to the above configuration, the program execution means holds musical tone signals for each output group obtained by executing the sound source processing program, and outputs each held musical tone signal at fixed output time intervals, for example, through 2 channels or 4 channels. It has musical tone signal output means for outputting to the D/A converter of the channel. In this case, the fixed output time interval is usually equal to the sampling period of the D/A converter, etc., but this time interval may be the same as the above-mentioned predetermined time interval, or the sound source processing program may be executed multiple times. When the musical tone signal for l samples is generated, the time interval can be set to one time interval corresponding to a plurality of predetermined time intervals.

In the above configuration, the program execution means can be configured to control the output ratio between the musical tone signal outputs of each output group based on the performance information.

The performance information in this case is, for example, information such as the velocity of an initial touch or the velocity of an aftertouch indicating a touch during a performance operation.

[For production]

In the present invention, the program storage means, address control means, data storage means, arithmetic processing means, and program execution means have the same configuration as a general-purpose microprocessor (microcomputer), and no dedicated sound source circuit is required. . Further, the musical tone signal output means has a configuration different from that of a general-purpose microprocessor, but is general-purpose in the scope of a musical waveform generator.

As a result, the circuit scale of the entire musical waveform generator can be significantly reduced, and even when integrated into an LSI, the manufacturing technology can be the same as that of ordinary microprocessors, and the yield of chips is improved, resulting in lower manufacturing costs. can be significantly reduced.

Note that since the musical tone signal output means can be constructed from a simple latch circuit, there is almost no increase in manufacturing costs due to the addition of this part.

In addition, when you want to change the sound source method, the polyphonic number, etc., you can simply change the sound source processing program stored in the program storage means.
It is possible to significantly reduce the development cost of a new musical waveform generator, and it is also possible for users to
It becomes possible to provide a new sound source method by using a card or the like.

The above effects are made possible because the present invention realizes the following program architecture and data architecture.

That is, the present invention realizes a data architecture in which musical tone generation data necessary for musical tone generation is stored on the data storage means. When the performance information processing program is executed, the corresponding musical tone generation data on the data storage means is controlled, and when the sound source processing program is executed, the corresponding musical tone generation data on the data storage means is controlled. A musical tone signal is generated. In this way, data communication between the performance information processing program and the sound source processing program is performed via musical tone generation data on the data storage means, and each program can access the data storage means based on the execution state of the other program. Since it is possible to perform the program without any involvement in the program, the two programs can essentially be configured as independent modules, resulting in a simple and efficient program structure.

In addition to the data architecture described above, the present invention normally executes a performance information processing program to operate keyboard keys and various setting switches, control demo performances, etc.
In contrast, a program architecture is realized in which a sound source processing program is executed at predetermined time intervals, and when the processing is completed, the program returns to the performance information processing program. As a result, the sound source processing program only has to forcibly interrupt the performance information processing program based on an interrupt signal generated at a predetermined time interval from the interrupt control means. There is no need to synchronize.

Furthermore, when the program execution means executes the sound source processing program, the processing time changes depending on the processing conditions, but this change can be completely absorbed by the musical tone signal output means. Therefore, a complicated timing control program for outputting musical tone signals to a D/A converter or the like is not required.

As described above, there is a data architecture in which the data is linked between the performance information processing program and the sound source processing program via musical sound generation data on the data storage means, and a sound source By realizing a program architecture for executing a processing program and further providing musical tone signal output means, sound source processing based on efficient program control can be realized with almost the same configuration as a general-purpose processor.

Further, the program execution means is configured to execute the performance information processing program and the sound source processing program by time-sharing processing corresponding to each sound generation channel. Therefore, the program execution means can generate musical tone signals for each sound generation channel by simply accessing and processing the corresponding musical tone generation data on the data storage means at each time division timing. Then, each sound generation channel is made to correspond to one of a plurality of output groups, such as stereo left and right channels, and for each output group, the musical tone signals generated by the sound generation channels included therein are mixed,
By generating musical tone signal outputs corresponding to each output group, the musical tone signals of each sound generation channel can be easily distributed and outputted to each output group.

Furthermore, if the output ratio between each musical sound signal output of each output group is controlled based on the performance information, it is possible to create musical sound effects such as moving the localization between the left and right channels by, for example, the key code or velocity value. It can also be added.

〔Example〕

Hereinafter, the present invention will be described in detail with reference to the drawings.

This disaster ship ■Emperor FIG. 1 is an overall configuration diagram of an embodiment of the present invention.

In the figure, first, the entire apparatus is controlled by a microcomputer 101. In particular, the microcomputer 101 executes not only the processing of control inputs for the musical instrument but also the processing of generating musical tones, and a tone generator circuit for generating musical tones is not required.

Switch section 10 consisting of a keyboard 102 and a pirate key 103
Reference numeral 4 denotes an operation input section of the musical instrument, and performance information input from the switch section 104 is processed by the microcomputer 101. Note that the details of the function key 103 will be described later.

The analog-converted musical tone signal generated by the microcomputer 101 is smoothed by a low-pass filter 105, amplified by an amplifier 106, and then emitted through a speaker 107. The power supply circuit 108 is connected to the microcomputer 1
01, supplies the necessary power to the low-pass filter 105 and amplifier 106.

For convenience, FIG. 2 is a block diagram showing the internal configuration of the microcomputer 101.

The control data/waveform ROM 212 stores tone control parameters such as a target value of an envelope value to be described later, tone waveform data for each sound source method, tone difference data, modulation waveform data, and the like.

Then, the command analysis unit 207 uses the control ROM 201
While sequentially analyzing the contents of the program, accessing each of the above data on the control data/waveform ROM 212,
Performs sound source processing using software.

A control ROM 201 stores a tone control program, which will be described later, and sequentially outputs program words (commands) at designated addresses from a ROM address control section 205 via a ROM address decoder 202. Specifically, the word length of each program word is, for example, 28 bits, and a part of the program word is input to the ROM address control unit 205 as the lower part (intra-page address) of the address to be read next. It is an address system. Of course, the normal program counter type C
It may be configured with PU.

The command analysis unit 207 analyzes the operation code of the command output from the control ROM 201, and sends control signals to each part of the circuit in order to execute the specified operation.

The RAM address control unit 204 specifies the address of a corresponding register in the RAM 206 when an operand of an instruction from the control ROM 201 specifies a register. The RAM 206 stores various musical tone control data for eight sound generation channels, which will be described later in FIGS. 6 and 7, as well as various buffers, which will be described later, and are used for sound source processing, which will be described later.

The ALU unit 208 and the multiplier 209 are controlled by the control jROM 31
If the instruction from is an arithmetic instruction, the command analysis unit 207
Based on instructions from , the former performs addition, subtraction and logical operations, and the latter performs multiplication.

The interwoven control section 203 supplies included signals to the ROM address control section 205 and the D/A converter sections 213 and 214 at fixed time intervals based on an internal hard timer (not particularly shown).

The switch unit 104 shown in FIG. 1 is connected to the input port 210 and the output port 211.

Various data read from the control ROM 201 or RAM 206 are sent to the ROM analog control unit 20 via the bus.
5, ALU section 208, multiplier 209, control data/waveform ROM 212, Left D/A converter section 213,
Right D/A converter section 214, input port 210
and output capo) 211. In addition, ALU section 2
08, Multiplier 209 and control data/waveform ROM 21
Each output of 2 is supplied to RAM 206 via a bus.

The left channel and right channel musical tone signals that have been subjected to sound source processing and obtained in the left buffer BL and right buffer BR (described later) in the RAM 206 are sent to the Left D/A converter section 2.
13 and Right D/A converter section 214, and the left channel analog musical tone signal R and the right channel analog musical tone signal R are sent to the speaker via the low-pass filter 105 and amplifier 106 shown in FIG. 107
The sound is emitted from. Note that both of these are stereo configurations.

Next, FIG. 3(b) shows the internal configuration of the Left D/A converter section 213 and the Right D/A converter section 214 in FIG. 1 (both have the same configuration).
, one sample data of a musical tone created by sound source processing is input to the latch 301 via the data bus. When a sound source processing end signal is input from the command analysis section 207 in FIG. 2 to the clock input of the latch 301, one sample of musical tone data on the data bus is latched into the latch 301 as shown in FIG. be done.

Here, since the time required for the sound source processing described above varies depending on the execution conditions of the software for sound source processing, the timing at which the sound source processing ends and the musical sound data is latched into the latch 301 is not constant. Therefore, as shown in FIG. 3(a), the output of the latch 301 is directly transferred to the D/A converter 303.
cannot be done manually.

Therefore, in this embodiment, as shown in FIG. 3(b), the latch 30
1 is further launched in the latch 302, and the musical tone signal is latched in the latch 302 by an included signal equal to the sampling clock interval outputted from the interlaced control unit 203 in FIG. I am trying to have it output to .

In this way, two latches are used to absorb changes in processing time in the sound source method, so a complicated timing control program for outputting musical tone data to the D/A converter is no longer necessary.

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

In this embodiment, the microcomputer 101 operates 35 (1
A series of processes from 2 to 5510 are repeatedly performed. Actual sound source processing is performed using interrupt processing. Specifically, at certain fixed time intervals, the data shown in Figure 5 (a
) is interrupted by the program being executed as the main flowchart, and based on the interrupt, a sound source processing program for creating 8-channel musical tone signals is executed.

When the processing is finished, the musical sound waveforms for 8 channels are added, and the left D/D in the microcomputer 101 is
A converter section 213 and Right D/A converter section 214
is output from. Thereafter, the interrupt state returns to the main flow. Note that the above-mentioned interrupt is periodically performed based on a hard timer in the included control unit 203 shown in FIG. This period is equal to the sampling period at the time of musical tone output.

The above is the general operation of this embodiment. Next, the overall operation of this embodiment will be explained in detail using FIG. 5.

The main flowchart in FIG. 5(a) shows the flow of processes other than sound source processing that are executed by the microcomputer 101 in a state where no interrupt is received from the interwoven control unit 203.

First, the power is turned on and the contents of the RAM 206 (see FIG. 2) in the microcomputer 101 are initialized (Sso+).

Next, each switch of the function key 103 (see FIG. 1) connected to the outside of the microcomputer 101 is scanned (S5D2), and the state of each switch is determined from the input port 2.
10 to the key buffer area in the RAM 206. As a result of the scanning, the function key whose state has changed is identified, and the corresponding function is processed (8503).
. For example, musical tone numbers are set, envelope numbers are set, and if rhythm performance is included as an additional function, rhythm numbers are set.

Thereafter, the keyboard keys being pressed on the keyboard 102 in FIG.
. 4) Key assignment processing is performed by identifying the changed key (S505).

This keyboard key processing is particularly relevant to the present invention, and will be described later.

Next, when a demo performance key (not shown) is pressed using the function key 103 (FIG. 1), demo performance data (sequencer data) is sequentially read out from the control data/waveform ROM 212 shown in FIG. , key assignment processing, etc. are performed (SS.6). Also, when the rhythm start key is pressed, the rhythm data is stored in the control data and waveform ROM.
212, and key assignment processing and the like are performed (SS.,).

After that, the timer process described below is performed (Ss
og). That is, the time value of the time data that is incremented by the interwoven timer processing (Ss+z) described later is determined, and the time value is determined as sequencer data for time control that is sequentially read out for demo performance control or time control sequencer data that is read out for rhythm performance control. By being compared with the rhythm data of S,. Time control is performed when performing the demo performance in step S 6 or the rhythm performance in step S 507.

In the sound generation process S509, pitch envelope processing is performed in which an envelope is added to the pitch of the musical tone to be generated, and pitch data is set in the corresponding sound generation channel.

Furthermore, one flow round preparation process is executed (Ss+o).

In this process, in the keyboard key processing of S5os, the state of the sound generation channel of the note number that started the key press is changed to "key pressed", or the state of the sound generation channel of the note number that started the key release is changed to "muted". etc. are performed.

Next, the interwoven process shown in FIG. 5(b) will be explained.

The included control unit 203 in FIG.
) When a program corresponding to the main flow is interrupted, the processing of the program is interrupted, and the process shown in Figure 5(b) occurs.
Execution of the included processing program is started. In this case, in the interwoven processing program, the fifth
The registers and the like to which writing is performed in the program of the main flow in FIG. Therefore, the process of saving and restoring the register and data, which is normally performed at the start and end of included processing, becomes unnecessary. As a result, the transition between the processing in the main flowchart of FIG. 5(a) and the included processing is quickly performed.

Subsequently, sound source processing is started in interwoven processing (Ssz). This sound source processing is shown in FIG. 5(C). As a result, musical waveform data accumulated for eight sound generation channels is obtained in buffer B, which will be described later, in RAM 206 in FIG.

Next, in 5512, interwoven timer processing is performed. Here, by taking advantage of the fact that the included processing in FIG. 5(b) is executed at a fixed sampling period,
The value of time data (not shown) on the RAM 206 (FIG. 2) is incremented. That is, by looking at the value of this time data, it is possible to know the passage of time. The time data thus obtained is used for time control in timer processing S5.8 of the main flow of FIG. 5(a), as described above.

Then, in 5513', the contents of the buffer area are transferred to the Left D/A converter section 213 and the Right D/A converter section 213.
It is latched by the latch 301 (see FIG. 3) of the D/A converter section 214.

Next, the operation of the sound source processing executed in step SS++ of the interwoven processing will be explained using the flowchart of FIG. 5(C).

First, left buffer B for waveform data addition in RAM 206
The areas L, right buffer BR, and flag F are cleared (Ss+3). Next, sound source processing is performed for each sound generation channel (S, ea~5521), and finally, when the sound source processing for the 8th channel is completed, the left buffer B
Waveform data obtained by adding eight channels to each of the L and right buffers BR is obtained. These detailed processes will be described later.

Next, FIG. 6 is a flowchart conceptually showing the relationship between the processes in the flowcharts of FIGS. 5(a) and 5(b) described above.

First, a certain process A (hereinafter the same applies to B, C1, . . . F) is performed (5601). This “processing” is shown in Figure 5 (a
), for example, "Function key processing j
, and "keyboard key processing". After that, interactive processing is entered, and sound source processing is started (S6 (+2
). As a result, in each of the left channel and the right channel, musical tone signals are obtained that are a collection of 8 sounding channels corresponding to ■ samples, and each of the left D/A converter sections 213
and R3ght is output to the D/A converter section 214.

After that, the process returns to some process B of the main flow (S60
3).

The above operations are repeated while sound source processing is performed for all eight sound generation channels (Sboa~5
611). This repeating process is continued while the musical tone is being generated.

Next, a specific example of the sound source processing executed at 3511 in FIG. 5(G) will be described.

In this embodiment, the microcomputer IO1 is responsible for the sound source processing for eight sound generation channels.

mentioned. The sound source processing data for eight channels is set in an area for each sound generation channel in the RAM 206 in FIG. 2, as shown in FIG.

In addition, the RA and M2O6 are provided with buffers BL and BR for waveform accumulation of the left channel and right channel as shown in FIG. Each area of a 1-bit flag F is reserved for allocation to channel processing.

In this case, each sound source method is set in each sound generation channel area in FIG. 7 as conceptually shown in FIG. 8 by operations described in detail later, and once the sound source method is set, Each control data is set from the control data/waveform ROM 212 in the data format of each sound source method as shown in FIG. Control data/waveform ROM21
The data format in No. 2 will be described later with reference to FIG.

Note that in this embodiment, different sound source systems are assigned to each sound generation channel, as will be described later.

In Table 1 showing the data format of each sound source method shown in FIG. 9, S is the sound source method No. which is a number for identifying the sound source method. The next A represents the address specified when waveform data is read out during sound source processing.
I, AI, and A2 are the integer part of the current address, and directly correspond to the address where the waveform data of the control data/waveform ROM 212 (FIG. 2) is stored. In addition, AF is the decimal part of the current address, and the control data and waveform ROM 212
It is used for interpolation of waveform data read from. The next AE represents an end address, and A represents a loop address. Also, the following P, ,P.

and P2 represents the integer part of the pitch data, and PF represents the decimal part of the pitch data. For example, P+=I, PF=0 indicates the pitch of the original sound, P, -2, PF=0 indicates the pitch one octave higher, and P+=0, PF=0.5 indicates the pitch below the ■octave. Each represents a pinch. The next Xp represents the previous sample data, and XN represents the storage of the next sample data. Further, D represents a difference value in size between two adjacent sample data, and E is an envelope value. Furthermore, 0 is an output value. In addition, as data particularly related to the present invention, L/R level Levl and p/f level L
There is ev2. For other various control data,
This will be explained later when explaining each sound source method.

In the present embodiment, when the main flow shown in FIG. Set in the channel area. Then, in the sound source processing for each channel in FIG. 5(C), which is executed as the sound source processing in the included processing in FIG. 5(C), the various control data set in the sound generation channel N area are used. Meanwhile, musical tone generation processing is executed. In this way, data communication between the main flow program and the sound source processing program is performed by the RA.
This is done via the control data (musical tone generation data) of the sound generation channel area on the M206, and access to the sound generation channel area in each program can be performed without regard to the execution status of the other program, so in reality both programs can be configured as independent modules, resulting in a simple and efficient program structure.

Hereinafter, sound source processing for each sound source method executed using such a data structure will be sequentially explained.

Nao, these sound source processing is done by microcomputer 10.
This is realized by the command analysis unit 207 of No. 1 interpreting and executing a sound source processing program stored in the control ROM 201. In the following, it is assumed that processing will be performed based on this premise unless otherwise specified.

First, in the flowchart of FIG. 5(C), each sound source process for each channel (any one of 3517 to 5524)
When entering the sound source, the sound source method number of the data in the data format (Table 1) shown in FIG. is determined to be executed.

Depends on PCM engineering 2. If the above sound source method No. indicates PCM method,
The PCM shown in the operation flowchart in Figure 10 below.
Sound source processing is performed according to the method. Each variable in the flow is each data in the PCM format of Table 1 in FIG. 9, which is stored in one of the sound generation channel areas in FIG. 7(a) on the RAM 206 (FIG. 2).

PCM on control data/waveform ROM 212 (Figure 2)
Among the addresses where waveform data is stored, the address where the waveform data to be processed currently is stored is indicated as (AIAF) shown in FIG. 12(a).

First, pitch data (P+, PF) is added to the current address (S+.01). This pitch data is the first
This corresponds to the type of key pressed on the keyboard 102 or the like in the figure.

Then, it is determined whether the integer part AI of the added address has changed (S+.02). If the judgment is NO,
DXA using the difference value which is the difference between the respective sample data XN and XP at address (AI+1) and A1 in FIG. 12(a).

The interpolation data value 0 corresponding to the decimal part AF of the address is calculated by the following calculation process (S1007).

Note that the difference value has been determined by sound source processing at interleaved timings before this time (see S 1 below).
006).

Then, sample data XP corresponding to the integer part A of the address is added to the interpolated data value 0, and new sample data 0 (
(corresponding to Xo in FIG. 12(a)) is obtained (Spo
o8).

Thereafter, this sample data is multiplied by the enherobe value E (Sl2O3) and is set as new sample data 0.

Furthermore, the sample data O is multiplied by L/R leheleve evl, and then multiplied by p/f leheleve ev2,
This is the final output value of 0 in that sounding channel (S 110.311111). These will be described later.

1 secured on the RAM 206 (Fig. 2)
Bit flag F (see Figure 7(b)) is 0 or 1
Depending on whether (S + o + z), the output 0 is accumulated in either the left buffer BL or the right buffer BR (
Stom3.5IOI, l) o Then flag F is incremented. Now, flag F is 5 in Figure 5 (C).
51:l is set to the initial value O. Therefore, Fig. 5 (C
) In the sound source processing of FIG. 10 in the first sound generation channel of S s + a, the flag F is O and the judgment of S I (+1□ is NO, the output 0 is accumulated in the left buffer BL, and S 1015 As a result, in the sound source processing in Figure 10 in the second sound generation channel of Figure 5(c) S5□5, flag F is 1 and S+o+
The determination of z is YES, the output O is accumulated in the right buffer BR, and the flag F returns to 0 again at 31015. As a result of repeating the above, each output 0 of the first, third, fifth, and seventh odd-numbered sounding channels is accumulated in the left Hanwha BL, and the outputs of the second, fourth, sixth, and eighth even-numbered sounding channels are accumulated. Each output O
is accumulated in the right buffer BR. That is, the odd-numbered sounding channels are the sounding channels for the left channel, and the even-numbered sounding channels are the sounding channels for the right channel. This will be detailed later.

After the above distribution is processed, the process returns to the main flow shown in FIG. Pitch data (PI, PF) is added to (AF) (S+oo+).

The above operation changes the integer part AI of the address (Sl2
It is repeated until O3).

During this time, the sample data Xp and the difference value are not updated, and only the interpolated data O is updated according to the address AF, and sample data Xo is obtained each time.

Next, when the integer part AI of the current address changes (S+ooz) as a result of adding the pitch data (PI, PF) to the current address (A1.AF) in S1001, it is determined whether the address AI has reached the end address AE. Or it is determined whether or not it exceeds (SIOf13).

If the determination is YES, the next loop process is performed. In other words, the address beyond the end address AE (A
tAE) is added to the loop address AL, and loop playback is started from the integer part A1 of the new current address obtained (S+oo4).

The end address A is the address on the control data/waveform ROM 212 (FIG. 2) where the last waveform sample data of the PCM waveform data is stored. Further, the loop address At is the address of the position at which the performer wants to repeat the output of the waveform, and the above-mentioned operation realizes the well-known loop processing in the PCM system.

If the determination in S 1003 is NO, the process in 51004 is not executed.

Next, the sample data is updated. here,
From the control data/waveform ROM 212 (FIG. 2), sample data corresponding to the newly updated current address A+ and the previous address (AI-1) are read out as XN and XP, respectively (Szoo5).

Furthermore, the previous difference value is updated to the updated difference value between XN and Xp (Sl2O3).

The subsequent operations are as described above.

In the manner described above, waveform data for one sound generation channel using the PCM method is generated.

Next, sound source processing using the DPCM method will be explained.

First, the operating principle of the DPCM system will be outlined using FIG. 12(b).

In the figure, a control data/waveform ROM 212 (second
The sample data Xp corresponding to address A in the figure) is
The address (not shown) immediately before address A (
This value is calculated from the difference value from the sample data corresponding to AI-1).

At address A+ of the control data/waveform ROM 212,
Since the difference value from the next sample data has been written, the sample data at the next address is determined by XP+D, and this is replaced as new sample data xP.

In this case, if the current address is AF as shown in the figure, the sample data corresponding to the current address AF can be found by X P + D X AF.

In this way, in the DPCM method, the difference value between the sample data corresponding to the current address and the next address is read from the control data/waveform ROM 212, added to the current sample data, and then read out from the control data/waveform ROM 212. By obtaining the data, waveform data is sequentially created.

When adopting this PCM method, when quantizing waveforms such as voices and musical tones where the difference value between adjacent samples is generally small, it can be quantized using a much smaller number of bits than the normal PCM method. It is clear that this can be done.

The operation of the above DPCM system will be explained using the operation flowchart of FIG. Each variable in the flow is stored in RAM
DPCM of Table 1 of FIG. 9 stored in any sound generation channel area of FIG. 7(a) on 206 (FIG. 2)
This is each data in the format.

Among the addresses where the DPCM difference waveform data on the control data/waveform ROM 212 is stored, the address where the data to be processed currently is stored is selected as the 12th address.
Let it be (A +, AF ) of (Fig. et al.).

First, pitch data (P
I, PF) are added (S++o+).

Then, it is determined whether there is a change in the integer part A+ of the added address (S1102). If the determination is NO, the interpolated data value O corresponding to the decimal part AF of the address is calculated using the difference (iD) at the address AI in FIG.
S 1114). Note that the difference value has been determined by the sound source processing at the interlaced timing before this time (51106 and S11□, which will be described later).

reference).

Next, sample data Xp corresponding to the integer part A1 of the address is added to the interpolated data value O, and new sample data O corresponding to the current address (AI, AF) (corresponding to χQ in FIG. 12(b)) is obtained (S+++s
).

Thereafter, this sample data is multiplied by the envelope value E (SII, 6), and new sample data 0 is obtained.

Subsequently, L/R level level ev to sample data O
1, p/f Level Lev2 multiplication (SII+7, S
z+g), and the resulting final output 0 of the sound generation channel is accumulated into the left buffer BL or right buffer BR based on the contents of the flag F (Sl119 to S11
2□) is 3101°~S in the PCM method in Figure 10.
The processing operation is exactly the same as that of 1015.

After the above distribution is processed, the process returns to the main flow shown in FIG. Pitch data (PI, PF) is added to AF) (SII.1).

The above operations are repeated until a change occurs in the integer part AI of the address.

During this time, the sample data Xp and the difference value are not updated, and only the interpolated data O is updated according to the address AF, and new sample data Xo is obtained each time.

Next, as a result of adding the pitch data (Pr, PF) to the current address (AI, AF) at Sth 01, when the integer part A1 of the current address changes (SIIoz), the address AI reaches the end address A. (SIIoa) If the determination is NO, the following loop processing from 5IIO4 to 3rd item 07 will
Sample data corresponding to the integer part A1 of the current address is calculated. That is, first, the variable old A1 (the 9th
The value before the integer part A1 of the current address changes is stored in the DPCM column of Table 1 in the figure. This is realized by repeating the processing of SI+06 or 311□3, which will be described later. The value of this old A is SI+0
S, while being sequentially incremented by 6. Old A at 7,
The differential waveform data on the control data/waveform ROM 212 (FIG. 2) instructed by is read out as D, and 5i
Lo! i is sequentially accumulated into sample data XP. Then, when the value of the old A1 becomes equal to the integer part A+ of the current address after the change, the J value of the sample data X becomes the value corresponding to the integer part AI of the current address after the change.

In this way, when the sample data Xp corresponding to the integer part AI of the current address is found, the determination of SI+04 is Y.
ES, and the process moves to the aforementioned interpolation value calculation process (SI114).

The above-described sound source processing is repeated for each interwoven timing, and when the determination in 51103 changes to YES, the next loop processing begins.

First, the address beyond the end address AE (At
AE) is added to the loop address AL, and the obtained address is set as the integer part A1 of the new current address (
Sth 08).

Thereafter, depending on how far the address has progressed from the loop address A, the operation of accumulating the difference value is repeated several times to calculate the sample data Xp corresponding to the integer part A1 of the new current address. . That is,
First, sample data X at the loop address AL where sample data
PL (see the DPCM column of Table 1 in Figure 9), and the old AI is the value of the loop address A (Sth ov), and the following 31110""'SI
The process of I+3 is repeated. That is, the value of old A1 is 31
11:? 5IIIO while being sequentially incremented by
ROM2 for control data and waveforms instructed by the old AI
The differential waveform data on 12 is read out as D, and the SI
At I+2, it is sequentially accumulated into sample data Xp.

Then, the value of old A1 is the integer part A1 of the new current address.
When it becomes equal to , the sample data XpO) value becomes a value corresponding to the integer part A+ of the new current address after the loop processing.

In this way, when sample data
).

In the manner described above, waveform data for one sound generation channel using the DPCM method is generated.

FM Next, sound source processing using the FM modulation method will be explained.

In the FM modulation method, hardware or software with the same content called operators is usually used, and musical tones are generated by interconnecting them according to certain connection rules called algorithms. In this embodiment, the FM modulation method is implemented using software.

Next, the operation of one embodiment when sound source processing is performed by two operators will be described using the operation flowchart of FIG. 13(a). The processing algorithm is shown in the same figure (b). In addition, each variable in the flow is stored in the RAM 206 (second
This is each data in the FM format of Table 1 in FIG. 9, which is stored in any of the sound generation channel areas in FIG. 7(a) above.

First, the modulator Operator 2 (OF2)
processing is performed. Regarding pitch processing, unlike the PCM method, interpolation is not performed, so only the integer address A2 is used. That is, the control data/waveform ROM 212 (
In FIG. 2), it is assumed that waveform data for modulation is stored at sufficiently narrow intermediate step intervals.

First, pitch data P2 is added to current address A2 (51301).

Next, feedback output FO2 is sent to this address A2.
is added as a modulation input to obtain a new address AMZ (513°2). The feedback output Fo2 is
5c3, which will be described later in the previous interwoven timing
This is obtained by executing the process in step 05.

Furthermore, the value of the sine wave corresponding to address AM2 (phase) is calculated. Actually, control data and waveform ROM2
The sine wave data is stored at 12, and the above address A
It is obtained by subtracting the sine wave data with H2 (S1303).

Subsequently, the sine wave data is multiplied by the envelope value E2 to obtain an output 02 (S13°4).

After this, the feedback level FL2 is applied to this output 02.
is multiplied and the feed hack output FO2 is obtained (S
1305). In this embodiment, this output FO2 is the operator 2 (O
F2).

Furthermore, modulation output M O2 is obtained by multiplying 02 by modulation level ML2 (S13
゜6). This modulation output Mo2 becomes a modulation input to operator 1 (OPI).

Next, the process moves to operator 1 (OPI). This process has the following features, except that there is no modulation input due to feedback output.
This is almost the same as the case of Operator 2 described above.

First, pitch data P1 is added to the current address AI of operator I (S13o7), and the above-mentioned modulation output Mo2 is added to this value to obtain a new address A old (S13011).

Next, the value of the sine wave corresponding to this address A, (phase) is read from the control data/waveform ROM 212 (SI3o9), multiplied by the envelope value E+ (SI310), and output O. Ru.

Subsequently, the L/R level Lev1 to the output O, p/
Multiplication of f level level ev2 (S1311.31312),
The resulting final output 0 of that sound channel,
The accumulation operation (31313 to S1316) to the left buffer BL or right buffer BR based on the contents of the flag F is exactly the same as the processing operation of S+o+o to S1015 in the PCM system in FIG.

This completes the FM modulation process for one sound generation channel.

By TM Next, sound source processing using the TM modulation method will be explained.

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

The above FM modulation method is e-A-sin (ωct+I(t)-sinω1It
) is the basic calculation formula. However, here, ωc1 is the carrier wave phase angle (carrier signal), sinω, L is the modulated wave phase angle (modulated signal), and I(t) is the modulation index.

On the other hand, in the phase modulation method called TM modulation method in this embodiment, e = A-fr (fc(t)+I(t) ・sinω
The basic calculation formula is m+, ). Here, h(t) is a triangular wave function, and is defined by the following function for each phase angle region (however, ω is an input).

1 block (ω)-2/π ・ω ・・(Area 2〇≦ω≦π/2) f□(ω)=−1+2/π(3π/2−ω)・・(Area: π/2≦ ω≦3π/2) b(ω)=-1+2/π(ω-3π/2)...(Area=
3π/2≦ω≦2π) In addition, fe is called a modified sine wave, and a control data/waveform ROM 212 (Fig. 2) storing different sine waveform data is carried for each phase angle region. phase angle ωc
This is the carrier signal generation function obtained by accessing at t. fc for each region of each phase angle is defined as follows.

fc(t) = π/2 sinωct・・(area:
0≦ωL≦π/2) fc(t)−π−π/2 sinω, t・・ (area: π≦ωt≦3π/2) fc(t)−2π+ W/2 sinωCt・・ (area : 3π/2≦ωct≦2π) (where n is an integer) In the TM modulation method, the modulation signal sinω is added to the carrier signal generated by the function fc(t) as described above with a modulation index of 1 (
The above-mentioned triangular wave function is modulated by the addition signal obtained by adding at the rate shown by t). This allows the modulation index I
If the value of (t) is O, a sine wave can be generated, and if the value of I (t) is increased, a very deeply modulated waveform can be generated.

Here, various signals can be used instead of the modulation signal sinω, and as described below, the operator's output from the previous calculation can be fed back at a certain feedback level, or the output of another operator can be input. You can also do so.

The sound source processing using the TM modulation method based on this principle is
This will be explained using the operation flowchart shown in FIG. 4(a). In this case, as in the case of the 13th FM modulation method, this is an example in which sound source processing is performed by two operators, and the processing algorithm is shown in FIG. 14 et al.). Further, each variable in the flow is each data in the 1M format of table 1 in FIG. 9, which is stored in one of the sound generation channel areas in FIG. 7(a) on the RAM 206 (FIG. 2).

First, the modulator Operator 2 (OF2)
processing is performed. Regarding pitch processing, unlike the PCM method, interpolation is not performed, so only the integer address A2 is used.

First, pitch data P2 is added to current address A2 (Sz.1).

Next, by the modified sine conversion fc, the above address A2
The modified sine wave corresponding to (phase) is the external memo IJ 11
6 (FIG. 1), and a carrier signal is generated as 0□ (S + aoz).

Next, the feedback output FO2 (31406) is added as a modulation signal to the above-mentioned carrier signal 02,
A new address is obtained and set to 02 (S+4o3)
. Feedback output F. 2 is obtained by executing S Idols processing, which will be described later, at the previous included timing.

Then, the value of the triangular wave corresponding to the above-mentioned addition address 02 is calculated. Actually, control data and waveform ROM2
12 (FIG. 2) stores the above-mentioned triangular wave data, and can be obtained by looking up the triangular wave data in a table in the a) palace 02 (Sxaoa).

Subsequently, the triangular wave data is multiplied by the envelope value E2 to obtain an output 02 (S1405).

After this, feed bank level FL2 is applied to this output 02.
is multiplied and the feedback output FO2 is obtained (S
1407). In this embodiment, this output FO2 is the operator 2 (O
F2).

Furthermore, 02 is multiplied by the modulation level M L 2 to obtain the modulation output Mo2 (S
e4゜7). This modulation output Mo2 becomes a modulation input to operator 1 (DPI).

Next, the process moves to operator 1 (OPI). This process has the following features, except that there is no modulation input due to feedback output.
This is almost the same as the case of Operator 2 described above.

First, the bit data P1 is added to the current address A of operator l (S1408), and the above-mentioned modified sine conversion is performed on the obtained value, so that the carrier signal becomes 0. (S +4ov).

Next, the above-mentioned modulation output M is added to this O+.

2 is added to create a new 01 (S141°), and this value 0. is converted into a triangular wave (SI411), and further multiplied by the envelope value E to obtain the output OI (SI411).
J+2).

E is multiplied (SI116) and the new sample data is then output to output 0 at L/R level Lev1, p/f.
Multiplication of level level eν2 (S□, 3, S□, 4), and accumulation operation (Se4ts ~31418) is exactly the same as the processing operation of S+o+o-S 1ot5 in the 10th PCM method.

This completes the TM modulation process for one sound generation channel.

With the explanation so far, PCM, DPCM, FM, TM
We have explained sound source processing using four methods. Among these, FM and TM are two modulation methods, and in the above example, the 13th
Although the processing by two operators based on the algorithm shown in FIG. 14(b) and FIG. 14(b) has been described, in actual sound source processing during performance, the number of operators may be larger and the algorithm may be more complex.

When playing an actual electronic musical instrument, a tone is assigned to each sound channel, and the musical tone signal generated based on that tone is distributed to the stereo left channel and right channel to be sounded. The specific operation of the process will be explained. This processing is realized as the processing operations of function key processing (S5.3), keyboard key processing S505, and sound generation processing 3509 in the main operation flowchart of FIG. 5(a).

Prior to the sound source processing that is performed for each sound channel as described above,
Input boat 210 (second
Based on the operation on the function keys 103 or the keyboard 102 shown in FIG. 1, which are connected to the operation panel of the electronic musical instrument via the 7(a)), the timbre parameters read from the control data/waveform ROM 212 (see FIG. 2) are set in the data formats of the various sound source systems described above (see FIG. 9). Based on the operating state of the keyboard 102, musical tone signals generated in each sound generation channel are distributed to a stereo left channel and a right channel, and are sounded.

First, FIG. 15(a) is a diagram showing an example of the arrangement of part of the function keys 103. As shown in FIG. In the figure, a part of the function keys 103 is realized as a tone specifying switch, and the performer selects the tones of ``Piano J, [Giku J, ... ``Koto'' of group A, ``Chu μ'' and ``clarinet'' of group B. ",...
The tone of "Chieguchi", the "violin" of Group C,
You can select the tone of "banjo", ... "harmonica". If a tone from group A is selected, the sound source processing shown in FIG. is performed, and if a C group tone color is selected, sound source processing is performed using the above-mentioned FM method.

In order to realize the above functions, RO for control data and waveform
M212 (see FIG. 2) stores various timbre parameters in a data structure as shown in FIG. 16.

That is, for each instrument corresponding to each timbre designation switch in FIG. 15(a), in group A, timbre parameters based on both the DPCM system and the TM system are stored,
ing. Each of these parameters is further divided into 4 & 11 timbre parameters of pL, fL, pR, and fR, which will be described later. Furthermore, in group B, tone parameters based on the PCM method are stored for each musical instrument. Furthermore, in group C, tone parameters based on the FM method are stored for each musical instrument.

Here, each timbre parameter set is stored in the data format of the various sound source systems shown in FIG.

Furthermore, in this embodiment, musical tones can be produced with different tones in the stereo left channel and right channel, and at the same time, the keyboard 10
Piano symbol p representing the strength corresponding to the key pressing speed in 2.
It is possible to produce musical tones with different tones between the forte symbol f and the forte symbol f, and it is also possible to output tones intermediate between the two.

In order to realize this function, a ROM for control data and waveform
In the data structure of the timbre parameters in Figure 16 stored in 21.2 (see Figure 2), p
4Mi timbre parameters of L, fL, pR, and fR are stored. pL is a tone parameter for the piano symbol p and for the left channel. fL is for the forte symbol f, and is a timbre parameter for the left channel.

PR is a tone parameter for the piano symbol p and for the right channel. fR is a timbre parameter for the forte symbol f and for the right channel.

A specific example of the process in which a tone is assigned to each sound generation channel using the tone parameters in Figure 16, and the musical tone signal generated based on that number is distributed to the stereo left channel and right channel and generated. The operations will be explained step by step.

First, FIG. 17 is a part of the operation flowchart of the s5o30 function key processing in the middle of the main operation flow chart of FIG. It is a flowchart.

First, it is determined whether or not there has been a change in the tone color designation switch shown in FIG. 15(a) (S1701). If no change occurs and the determination is NO, no special processing is performed.

There is a change in the tone designation switch and the judgment of 317°1 is Y.
If it is ES, then it is determined whether a C group tone has been designated (S1702).

When the C group tone is specified, the FM corresponding to the specified instrument of the C group stored in the control data/waveform ROM 212 (FIG. 2) as shown in FIG.
The tone parameters according to the method are stored in the RAM 206 (Figure 2).
It is set (S+□.3) in each of the upper sound generation channel areas (see N in FIG. 7(a)).

That is, first, a sound source system number indicating the FM system is set in the leading area S (see FIG. 9) of each sound generation channel area. Subsequently, various parameters corresponding to the currently designated timbre are set in the second and subsequent areas of each sound generation channel area.

Here, in the first sound channel area (ch 1 ),
A timbre parameter pL for the piano symbol p and for the left channel is stored, and a timbre parameter pR for the piano symbol p and for the right channel is stored in the second sound generation channel area (ch2). 3 sound channel area (ch3
) stores the timbre parameter fL for the forte symbol f and for the left channel, and the fourth sounding channel area (
ch4) stores a tone parameter fR for the forte symbol f and for the right channel. Further, the fifth to eighth sound generation channel areas (ch5 to ch8) respectively store the same tone parameters as those of the first to fourth sound generation channel areas.

By using each sound generation channel to which timbre parameters have been assigned in this way, in this embodiment, as will be described later, one key press operation can be performed using four channels, the first to fourth sound generation channels, or , 4 of the 5th to 8th pronunciation channels
A sound generation instruction is given to any four channels simultaneously, and one musical tone signal is generated by the four channels. In other words, the number of notes that can be produced simultaneously is two.

On the other hand, if the timbre of the C group is not designated and the determination in 5I702 is No, it is determined whether the timbre of the B group is designated (S+)o4).

When the B group tone is specified, the B group stored in the control data/waveform ROM 212 as shown in FIG.
Tone parameters based on the PCM method corresponding to the designated instruments of the group are set in each sound generation channel area (see FIG. 7(a)) on the RAM 206 (S+□.5).

That is, first, the sound source system number indicating the PCM system is set in the leading area S (see FIG. 9) of each sound generation channel area. Subsequently, various parameters corresponding to the currently designated timbre are set in the second and subsequent areas of each sound generation channel area. In this case, the first to
When C group tones are specified in the fourth sound channel area and the fifth to eighth sound channel areas (S170
Similarly to 3), the tone parameters pL, pR, fL, and fR are stored.

If the tone of group B is not specified and the determination in S1704 is NO, that is, if the tone of group A is specified, the tone parameters of group A are stored in RAM2.
The settings for each sound generation channel area above 06 are not performed by function key processing, and the flow part of FIG. 17 is ended as it is. Setting of the tone color parameters of the A group to each sound generation channel area on the RAM 206 is performed in the keyboard key processing described below.

Next, the keyboard key processing (3) of the main operation flowchart in FIG. 5(a) when playing an actual electronic musical instrument
505) will be explained.

First, a first embodiment of keyboard key processing when a key is pressed will be described.

In the first embodiment of the keyboard key processing when a key is pressed, first, it is determined whether or not a tone of group A is currently designated (31so+).

If the timbre of the A group is specified, as described above, the timbre parameters of the A group have not yet been assigned to each sound generation channel, so that processing is performed. In this case, the sound source system set for each sound generation channel is automatically switched depending on the speed at which keys on the keyboard 102 are pressed, that is, the velocity. In this case, M I D I (Mus
ical Instrument Digital In
64, which is 1/2 of the maximum value of the standard (127)
The DPCM method is used for fast key presses with a velocity value of 64 or more, and the TM method is assigned for slow key presses with a velocity value of 64 or less. It will be done.

That is, in S11102, it is determined whether or not the herocity of the key determined to be a "pressed key" in the keyboard key import process of 3504 in the main operation flowchart of FIG. 5(a) is 64 or more. It will be judged.

Note that this velocity value of 64 is based on the MIDI standard mp (
equivalent to a mezzo piano).

SIB if the velocity value is 64 or higher. Decision 2 is YES
In this case, control data and waveform ROM 212 (Figure 2)
Among the tone parameters corresponding to the designated instruments of the A group stored as shown in FIG. reference) is set to (
S18°3). That is, first, a sound source system number indicating the DPCM system is set in the leading area S (see FIG. 9) of each sound generation channel area. Subsequently, various parameters corresponding to the currently designated timbre are set in the second and subsequent areas of each sound generation channel area. In this case, the 1st to 4th sound channel areas and the 5th to 8th sound channel areas have pL, PR', fL, as in the case where C group tones are specified (517oz in Figure 17).
, fR are stored.

If the herocity value is less than 64 and the determination of 5IliO2 is NO, the first
Among the tone color parameters corresponding to the designated instruments of group A stored as shown in FIG. 6, the tone parameters according to the TM method are set in each sound generation channel area on the RAM 206 (S+5o4).

That is, first, the first area S (9th
(see figure), the sound source method number indicating the TM method is set. Subsequently, various parameters corresponding to the currently designated timbre are set in the second and subsequent areas of each sound generation channel area. In this case as well, pL
, pR, fL, and fR are stored.

Through the processing up to this point, setting of tone parameters to each sound generation channel is completed. Next, the steps 1 to 1 in FIG. 5(C)
Processing of the fourth sound generation channel (S513) or the fifth to eighth sound generation channels
Next, we move on to processing for mixing the musical tone outputs of four channels obtained by processing the sound generation channels to obtain one musical tone output. This is indicated by 31[105-311108 in FIG.

Now, suppose that, as part of the keyboard key processing 3505 in FIG. 5A, a sound generation instruction is given to, for example, the first to fourth sound generation channels. As mentioned above, the first sounding channel area stores the tone parameters pL for the piano symbol p and for the left channel, and the second sounding channel area stores the tone parameters pL for the piano symbol p and for the right channel. The timbre parameter pR for the forte symbol f and the left channel timbre parameter fL is stored in the third sound channel area.
is stored, and the timbre parameter fR for the forte symbol f and for the right channel is stored in the fourth sound generation channel area.

In this embodiment, first, depending on the position on the keyboard when a key on the keyboard 102 is pressed, that is, the range of the musical tone, whether to increase the output of the left channel tone or the right channel tone is determined. It can be controlled as follows.

Now, in SI1+05, the output level for each sound generation channel is set in a register (not shown) in the RAM 206. For example, when a key in a low range is pressed, the output level of the first and third sound generation channels, to which the tone parameters for the left channel are set, is increased, and the tone parameters for the right channel are set. The output levels of the second and fourth sound generation channels are reduced, and settings are made so that the two add up to a value of 1. The level settings in this case are shown in Figure 19 (
It is done according to the characteristics of a). In the figure, the characteristic indicated by "Lj" is the output level straight line for the left channel, and the characteristic indicated by "rR" is the output level straight line for the right channel.

Next, at 51806, the output level corresponding to each sound generation channel set as above is stored in the RAM 206.
(Figure 2) L/R level of each sound channel area above
vl (see Figure 9).

The L/R level Levl set in this way is 31010 in FIG. 10 and 51117 in FIG. 11 for each sound source process described above.
, 31311 in Fig. 13 and 31413 in Fig. 14, by multiplying the output 0 of each sound generation channel,
It is possible to obtain a tone and localization closer to the left channel or the right channel depending on the range of the pressed key.

In addition to the above-mentioned localization of the left and right channels, in this embodiment, the output of the tone for the piano symbol p is increased or the output of the tone for the forte symbol You can increase the tone output or control it as follows.

For example, if the key pressing speed is fast, increase the output level of the first and second sound generation channels in which the tone parameters for the piano symbol p are set at S Il+(+7,
Settings are made such that the output levels of the third and fourth sound generation channels, in which the tone color parameters for the forte symbol f are set, are made small, and the sum of the two becomes 1. Level setting in this case is performed according to the characteristics shown in FIG. 19(b). In the figure, the characteristic indicated by 'PJ' is the output level straight line of the tone for the piano symbol p, and the characteristic indicated by "f" is the output level straight line of the tone for the forte symbol f.

Next, at 311108, the output level corresponding to each sound generation channel set as above is stored in the RAM 20.
6 (second time) The p/f level of each sound generation channel area is set to level ev2 (see FIG. 9).

The p/f level Lev2 set in this way is shown in FIG. 1O S +o++ and FIG.
111, 3131□ in Fig. 13 or Sxx in Fig. 14, by multiplying the output O of each sound generation channel,
Piano symbol p or forte symbol f depending on key pressing speed
You can get a closer tone.

The processing operations up to this point are performed in exactly the same manner for the fifth to eighth sound generation channel areas.

A specific example of tone color setting and level setting according to the first embodiment of the above key press key processing is shown in FIG. 24(a).

In the example in the same figure, the PIAN of group A in FIG. 15(a)
This is the case when the O tone color designation key is pressed. And the first ~
Since the keys simultaneously assigned to the fourth sound channel have a velocity of 64 or less (value 20), the TM method is assigned, and based on the characteristics shown in FIG. 19(b), the piano symbol p/f level Lev2 of the first and second sound generation channels where tone parameters for P are set
becomes a large value such as 0.8426, and the p/f level ev2 of the 3rd and 4th pronunciation chaor, where the tone parameter for the forte symbol f is set, is 0.1574.
, and the sum of both is l. Since the scale indicated by the key code is G3 (key code number 19, see Figure 15 (a)) and is closer to the low range, the tone parameters for the left channel are set based on the characteristics shown in Figure 19 (a). The L/R level level evl of the 1st and 3rd sounding channel that is set is a large value such as 0.6985, and the L/R level level of the 2nd and 4th sounding channel where the tone parameter for the right channel is set. Kanae ev
l is a small value such as 0.3015, and the sum of both is 1.

On the other hand, the pressed state of the keys that are simultaneously assigned to the 5th to 8th sound generation channels is the exact opposite of the pressed state of the keys that are assigned to the 1st to 4th sound generation channels. The velocity is high (value 100) and the scale is high (C7,
Key code number 60 (see FIG. 15(b)). Therefore,
As shown in FIG. 24(a), the DPCM method corresponding to large velocity is assigned to the fifth to eighth sounding channels, and based on the characteristics shown in FIG. The p/f level Lev2 of the first and second sound generation channels for which tone parameters are set is 0.212
6, and the p/f level e■2 of the 3rd and 4th pronunciation channels, where the tone parameters for the forte symbol f are set, have a large value of 0.7874. Based on the characteristics shown in FIG. 19(a), the L/R level Levl of the first and third sound generation channels for which the timbre parameters for the left channel are set is 0.
The L/R level Levl of the second and fourth sound generation channels to which the tone parameters for the right channel are set becomes a small value such as 0.9523.

Next, a second embodiment of keyboard key processing when a key is pressed will be described.

In the second embodiment of keyboard key processing when a key is pressed, the above-mentioned first
Contrary to the case of the above embodiment, when it is determined that the tone of group A is currently specified (SZ□.1) and the tone parameters of group A are assigned to each sound generation channel, the keyboard 102 The sound source system set for each sound generation channel is automatically switched depending on the range of the key pressed above.

That is, in 32202, whether or not the value of the key code of the key that was determined to be a "pressed key" in the keyboard key import process of 3504 in the main operation flowchart of 5th [D(a)] is 31 or less. (See Figure 15) is determined.

82□ if the key code value is 31 or less. Decision 2 is YES
In this case, control data and waveform ROM 212 (Figure 2)
Among the tone parameters corresponding to the designated instruments of the A group stored as shown in FIG. reference) is set to (
32203). This is the same as the 3 180:l processing in the first embodiment shown in FIG.

The value of the key code is thicker than 31 (determination of 32202 is N
In the case of O, the 16th
Among the tone color parameters corresponding to the designated instruments of the A group stored as shown in the figure, the tone color parameters according to the TM method are set in each sound generation channel area on the RAM 206 (32204).

This is 3 1804 in the first embodiment of FIG.
The process is the same as that of .

Next, the musical tone outputs of each of the four channels obtained by the processing of the first to fourth sound generation channels (3513) or the processing of the fifth to eighth sound generation channels in FIG. 5(C) are mixed to produce one musical tone. Let's move on to the process for obtaining output. This is S2 in Figure 20.
□. 5 to indicated by szzog. Here, contrary to the first embodiment shown in FIG. 18, the output of the left channel tone is increased or the right channel tone output is increased depending on the pressing speed of the keys on the keyboard 102, that is, the velocity. The output of the tone of the piano symbol P is increased or the forte symbol f is controlled by the key code of the pressed key.
Controls whether to increase the tone output.

For example, if the velocity value is small (key pressing speed is slow), in S2205, the output level of the first and third sounding chaffle, to which the tone parameters for the left channel are set, is increased, and the right Settings are made such that the output levels of the second and fourth sound generation channels for which channel tone parameters are set are reduced, and that the sum of the two becomes 1. In this case, the level setting is
This is done according to the characteristics shown in FIG. 23(a). In the same figure, “L
” is the output level of the left channel, rR
The characteristic indicated by `` is a straight line in the output level of the right channel.

Next, in the 3220&, the output level corresponding to each sound generation channel set as above is stored in the RAM 206.
(Figure 2) L/R level of each sound channel area above
vl (see Figure 9).

The L/R level level evl set in this way is 31010 in Fig. 10 and SII+7 in Fig. 11 of each sound source processing described above.
, 5I311 in FIG. 13 or 31413 in FIG. 14, by multiplying the output 0 of each sound generation channel, a timbre and localization closer to the left channel or right channel depending on the key depression speed, that is, the velocity value can be obtained.

On the other hand, for example, if the pitch range of the pressed key is low, that is, the value of the key code is small, in S2□o7 the output levels of the first and second sound generation channels to which the tone parameters for the piano symbol p are set are changed. and the third and fourth, where the timbre parameters for the forte symbol f are set.
Settings are made such that the output level of the sound generation channel is reduced and the two add up to a value of 1. Level setting in this case is performed according to the characteristics shown in FIG. 23(b). In the same figure, the characteristic indicated by 'PJ is the output level straight line of the tone for the piano symbol p, and the characteristic indicated by rfJ is the output level straight line of the tone for the forte symbol f.

Next, in S22011, the output level corresponding to each sound generation channel set as above is stored in RAM2.
The p/f level level ev2 (see FIG. 9) of each sound generation channel area on 06 (FIG. 2) is set.

The p/f level Lev2 set in this way corresponds to S 1o++ in FIG. 10 and 311 in FIG. 11 for each sound source process described above.
18. In Fig. 13 SI:I+2 or Fig. 14 5I414, by multiplying the output O of each sound generation channel, the tone closer to the piano symbol p or the forte symbol f according to the range of the pressed key is created. can get.

The processing operations up to this point are performed in exactly the same manner for the fifth to eighth sound generation channel areas.

A specific example of tone color setting and level setting according to the second embodiment of the above key press key processing is shown in FIG. 24(b).

In the example in the same figure, the PIAN of group A in FIG. 15(a)
This is the case when the O tone color designation key is pressed. And the first ~
The pressed state of the keys simultaneously assigned to the fourth sound generation channel is assigned the DPCM method because the scale is as low as 03 (key code number 19), and based on the characteristics shown in FIG. 23(b), The p/f level Lev2 of the first and second pronunciation channels has a large value such as 0.6985, and the timbre parameter for the forte symbol f has been set. and the p/f level Lev2 of the fourth sound generation channel has a small value of 0.3015, and the sum of both is 1. Since the velocity value is 100, which is large, the L/R level evl of the first and third sound generation channels, in which the timbre parameters for the left channel are set, is 0.2126 based on the characteristics shown in FIG. 23(a). The L/R level evl of the second and fourth sound generation channels to which the tone parameters for the right channel are set becomes a small value such as 0°7874,
Well, the sum of both is 1.

On the other hand, the pressed state of the keys that are simultaneously assigned to the 5th to 8th sound generation channels is the exact opposite of the pressed state of the keys that are assigned to the 1st to 4th sound generation channels. The range is high < (C7, key code number 60, Figure 15 (b)
), the velocity is small ((!100). Therefore,
As shown in FIG. 24(b), the TM method corresponding to the high range is assigned to the fifth to eighth sound generation channels, and the tone for piano symbol P is assigned based on the characteristics shown in FIG. 23(b). The ρ/f level ev2 of the first and second sound generation channels where the parameters are set becomes a small value such as 0.0477, and the ρ/f level ev2 of the third and fourth sound generation channels where the tone parameters for forte symbol f are set. The p/f level Lev2 becomes a large value such as 0.9523, and on the other hand, the L of the first and third sound generation channels where the tone parameters for the left channel are set based on the characteristics shown in FIG. 23(a). /R level Level ev 1 is a large value such as 0.8426, and the L/R of the second and fourth sound generation channels where the tone parameters for the right channel are set.
The level level evl becomes a small value such as 0.1574.

As described above, after the level control is performed for each sound generation channel by the keyboard key processing at the time of key depression shown in the first or second embodiment, the above-mentioned steps 31012 to S10 in FIG. 10 are performed.
15, Figure 11 31119-51122, Figure 13 31
313~S 1316 or Figure 14 S +aI5~31
18, the output 0 of the odd-numbered sounding channels is accumulated in the left buffer BL, and the output O of the even-numbered sounding channels is accumulated in the right buffer BR.

Then, the sound source processing for 8 sound generation channels of S 5ea to S 521 is completed (FIG. 5 (1)) in FIG. 5(C).
After returning from the included processing to the processing of the main flow of FIG. 5(a), in the sound generation processing of 35f+9 of FIG. Output processing to the device section 214 is performed. A part of this process is shown in FIG. That is,
Various effect processing (for example, LFO
After processing (effect processing, etc.), left/N7FyBL
The value of is latched in the latch 3o1 (Fig. 3) in the Left D/A converter section 213 (Fig. 2) (S21
1111). Also, the value of the right buffer BR is
The latch 301 (in the D/A converter section 214 (Fig. 2)
(Fig. 3) is latched (Szooz).

After this, the latch 30
The musical tone signal latched at 1 is passed through the latch 302 to D/
The signal is output to the A converter 303, where it is converted into an analog tone signal, and the sound is emitted from the speaker 107 (both having a stereo configuration) via the low-pass filter 105 and amplifier 106 shown in FIG.

Next, when the pressed key of the keyboard 102 is released, as part of the operation process of the keyboard key process of s sos in the main operation flowchart of FIG. 5(a),
The operation flowchart of FIG. 21 is executed. That is, in the keyboard key import processing S504 of FIG. 5(a), "
1st to 4th keys to which keys determined to be released are assigned.
Necessary tone parameters in the sound generation channel area (see FIG. 7(a)) on the RAM 206 (FIG. 2) corresponding to the sound generation channels or the four sound generation channels of the fifth to eighth sound generation channels are initialized. However, the timbre parameter data itself remains in each sound generation channel area without being erased. Thereafter, when the four sound generation channels are assigned again by pressing a key, the above-described keyboard key processing at the time of key pressing shown in FIG. 18 will be executed.

In the embodiment of the present invention described above, the timbre parameters set for each sound channel correspond to the piano symbol p and forte symbol f of the dynamic symbol. It is not limited to the first tone, the second tone in the -i sense.
It could also be something like the tone of. Further, the number is not limited to two types, but may be a plurality of types, such as pianissimo symbol pp, piano symbol p, forte symbol f, fortissimo symbol ff, or the first to fourth tones.

Further, although the timbre mixing level is controlled using velocity and range information, it may also be controlled using various performance information such as aftertouch.

Further, although the D/A conversion output is a two-channel stereo output, it may be other than that, for example, a four-channel stereo output.

〔Effect of the invention〕

According to the present invention, there is no need for a dedicated sound source circuit;
It becomes possible to have a general-purpose processor configuration. As a result, the overall circuit scale of the musical waveform generator and equipment can be significantly reduced, and even when integrated into an LSI, the manufacturing technology is the same as that used for ordinary microprocessors, and the chip yield is improved. It becomes possible to significantly reduce costs. Note that since the musical tone signal output means can be constructed from a simple round-circuit circuit, there is almost no increase in manufacturing costs due to the addition of this part.

In addition, when you want to change the sound source method, the polyphonic number, etc., you can simply change the sound source processing program stored in the program storage means.
It becomes possible to significantly reduce the development cost of a new overall musical waveform device, and it also becomes possible to provide users with a new sound source system using, for example, a ROM card.

In this case, a data architecture that links data between the performance emotion processing program and the sound source processing program via musical sound generation data on the data storage means;
By realizing a program architecture in which the sound source processing program is executed at predetermined time intervals in relation to the performance information processing program, there is no need for processing to synchronize both processors, making it possible to greatly simplify the program. becomes.

Furthermore, since any changes in the processing time of sound source processing depending on the sound source method can be absorbed by the musical sound signal output means,
Another effect is that a complicated timing control program for outputting musical tone signals to a D/A converter or the like is not required.

In particular, in the present invention, musical tone signals generated by sound generation channels included in each output group are mixed to generate a musical tone signal output corresponding to each output group.
It becomes possible to easily distribute and output musical tone signals of each sound generation channel to each output group.

In addition, if the output ratio between musical tone signal outputs of each output group is controlled based on performance information, the stereo location can be moved between each output group based on performance information such as key chord or velocity values. It is also possible to easily add musical sound effects such as

[Brief explanation of the drawing]

FIG. 1 is an overall configuration diagram of an embodiment according to the present invention, FIG. 2 is an internal configuration diagram of a microcomputer, and FIG. 3(a) is a configuration diagram of a conventional D/A converter section. [Yes]) is
A configuration diagram of the D/A converter section according to the present embodiment. Fig. 4 is a timing chart in D/A conversion. Fig. 5 (a) to (C) are overall operation flowcharts of this embodiment. Fig. 6 is a conceptual diagram showing the relationship between the main operation flowchart and included processing, Figures 7(a) and (b) are diagrams showing the data structure of the RAM, and Figure 8 is a diagram showing the sound source processing method for each sound channel. Conceptual diagram for selection. Figure 9 is a configuration diagram of data formats for each sound source method on the RAM. Figure 10 is an operation flowchart of sound source processing using the PCM method. Figure 1 is a sound source processing using the DPCM method. The operation flowchart of FIG. 12 (a) and (b) shows the difference value and the current address A.
Principle explanatory diagram when calculating the interpolated value Xo using v, 1st
Figure 3 (a) is an operation flowchart of sound source processing using the FM method, Figure 13 (c) is a diagram showing an algorithm for sound source processing using the FM method, and Figure 14 (a) is an operation flowchart of sound source processing using the TM method. Flowchart, Fig. 14 0)) is a diagram showing the algorithm of sound source processing using the TM method, Fig. 15 is a diagram showing an example of the configuration of the switch section, and Fig. 16 is a diagram showing the data configuration of timbre parameters. , FIG. 17 is an operational flowchart of function key processing, FIG. 18 is an operational flowchart of the first embodiment of keyboard key processing when a key is pressed, and FIG. 19 is a first example of keyboard key processing when a key is pressed. FIG. 20 is an operation flowchart of the sound production process, FIG. 21 is an operation flowchart of the keyboard key processing example when a key is released, and FIG. 23(a) and 23(b) are operation flowcharts of the second embodiment of keyboard key processing when a key is pressed. FIGS.
FIGS. 24(a) and 24(b) are diagrams illustrating the relationship between heaviness, musical scale, and level in the embodiment of FIG. 101...Microcomputer, 102...Keyboard, 103...Function key, 104...Switch section, 105...Low pass filter, 106...Amplifier, 301. 303 - Speaker, - Power supply circuit, - Control wholesale ROM, - ROM address decoder, - Included control section, - RAM address control section, - ROM address control section, - RAM.・Command analysis section, ・ALU section, ・Multiplier, ・Input port, ・Output port, ・Control data/waveform ROM, -Left D/A converter section, ・Right D/A converter section, 02...・Latch, ・D/A converter. C door D

Claims (1)

[Scope of Claims] 1) Program storage means for storing a performance information processing program for processing performance information and a sound source processing program for obtaining musical tone signals, and address control means for controlling the address of the program storage means. and a data storage means for storing musical tone generation data necessary for generating a musical tone signal for each sound generation channel; and an arithmetic processing means; while controlling the address control means, the data storage means, and the arithmetic processing means, A 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 corresponding musical sound generation data on the data storage means. The means transfers control to the sound source processing program at predetermined time intervals to execute it, and executes the performance information processing program again after the program is finished, and when the sound source processing program is executed, the control is executed for each of the sound generation channels. , performing time division processing based on the musical tone generation data on the data storage means to generate a musical tone signal corresponding to each of the sound generation channels, and making each sound generation channel correspond to one of a plurality of output groups; program execution means for generating a musical tone signal output for each output group by mixing musical tone signals generated by sound generation channels included in each output group; and a program execution means for generating a musical tone signal output for each output group; Musical sound waveform generation characterized by having a musical sound signal output means for holding musical sound signals for each of the output groups obtained by the execution and outputting each of the held musical sound signals at fixed output time intervals. Device. 2) The program execution means includes an interrupt control means that generates an interrupt signal at the predetermined time interval, and the program execution means is configured to generate the interrupt signal from the interrupt control means while executing the performance information processing program. The performance information processing program is interrupted at the timing when the performance information processing program is interrupted, control is transferred to and executed by the sound source processing program, and after the termination, the interruption is canceled and execution of the performance information processing program is resumed. The musical sound waveform generator according to claim 1. 3) The musical sound waveform generator according to claim 1 or 2, wherein the plurality of output groups correspond to each stereo output channel. 4) The musical sound waveform generator according to claim 1, wherein the program execution means controls an output ratio between musical tone signal outputs of each output group based on the performance information. 5) The musical sound waveform generator according to claim 4, wherein the performance information used for controlling the output ratio is information indicating a pitch range. 6) The musical sound waveform generating device according to claim 4, wherein the performance information used to control the output ratio is information indicating a touch during a performance operation.