JPH02179692A - Processor for electronic musical instrument - Google Patents

Abstract

PURPOSE:To reduce the circuit scale to reduce the cost by generating musical sounds by the program control of a microcomputer. CONSTITUTION:Information is inputted from a switch part consisting of function keys and keyboard keys and is stored in a RAM 34. A program to process the control input and a program to generate musical sounds are stored in a control ROM 31, and the instruction in an address designated by a ROM address control part 39 is outputted. When the operand of the instruction from the ROM 31 designates a register, a RAM address control part 33 designates an address of the RAM 34. The RAM 34, an adder/subtractor, and a multiplier 36 constitute an operation circuit. Envelope data and musical sound waveform data are stored in a ROM 37 for control data as well as waveform. An operation analyzing part 38 decodes the instruction from the ROM 31 to send a control signal to each part of the circuit. The control part 39 sends the control signal at intervals of a certain time by an interrupt control part 40 having a timer to generate musical sounds. Generated musical sounds are converted to an analog signal and are outputted to the external.

Description

DETAILED DESCRIPTION OF THE INVENTION [Technical Field of the Invention] The present invention relates to a processing device for an electronic musical instrument, and more particularly to the structural architecture of a processing device for an electronic musical instrument.

[Prior art and its problems] In recent years, electronic musical instruments have been computerized.

However, the part related to the generation of musical tones, which requires large amounts of high-speed data calculation, is performed by dedicated hardware called a sound source circuit, and the microcomputer inputs control inputs to the musical instrument from the control input cap 1 and the console panel. It merely processes MIDI and other external control inputs, inputs from internal or external performance memories, etc.) and transfers appropriate commands to the tone generator circuit.

The structure of the sound source circuit differs depending on the method of musical tone synthesis, but the circuit size of each sound source method is large.
Typically, it is about twice the circuit scale of a microcomputer (central processing unit).

- As an example, there is a microcomputer 101 that controls a PCM sound source lOO, whose block diagram of a PCM sound source type is shown in FIG.
The signal is then sent to the PCM sound source 100. Commands from the microcomputer 101 are set to each part of the sound source via the sound source command analysis section 102.

For example, at the start of pronunciation, information is set in the following procedure.

(a) Send an address (usually consisting of a start address, end address, and loop address) to the waveform storage device 107 containing the waveform to be generated. These addresses are set in address control section 104.

(b) Send pitch data of the musical tone to be produced. The pitch data is set in the pitch control section 105.

(C) En<O-7'7''-Y is sent and set in the envelope control section 106.

(d) Turn on channel control (channel ON1
0FF set in the control unit 103).

In the case of a polyphonic sound source, these data must match the channel numbers, and each part of the sound source 100 must operate in a time-division manner.When 0 or more data is set, the PCM sound source 100 generates musical sounds as follows. generate. At the corresponding channel time, the address control unit 104 reads from the waveform storage device 107 the waveform data (the previous waveform value and the immediately following waveform value) located at the two adjacent addresses closest to the cumulative result of the pitch data from the pitch control unit 105. , this waveform data is sent to the waveform processing section 108, where the difference between the immediately preceding waveform value and the immediately following waveform value is calculated. This difference and the previous waveform value are sent to the multiplication circuit 109, where the difference between adjacent waveform values is added to the decimal part FD of the address of the waveform storage device.
(In the figure, given from the pitch control unit lO5) is multiplied by
An interpolated value is obtained by adding the previous waveform value to it, and an envelope value E generated by the envelope control unit lO6 is added to this interpolated value.
Multiply by D to obtain the instantaneous value of the tone waveform of the channel. This instantaneous value is accumulated for all channels by an adder 110, and the result is sent to a D/A converter 111 to become an analog tone signal.

As can be seen from this example, the hardware of the sound source circuit has the problem that arithmetic circuits and storage devices for temporarily holding data are required at various processing stages, making one circuit large-scale. Furthermore, the structure of a specific tone circuit only realizes musical tone synthesis using a specific sound source method and a specific polyphonic number, and even changing the polyphonic number requires significant circuit changes and additions. Furthermore, it is necessary to design a set of commands sent from the microcomputer to the sound source circuit in accordance with the sound source, which requires a great deal of time and effort to develop a sound source control program.

[Object of the Invention] Therefore, the object of the present invention is to provide a processing device for an electronic musical instrument with a new structure that enables musical tones to be generated under program control of a microcomputer and does not require dedicated tone generator circuit hardware. .

[Structure 1 of the Invention] According to the present invention, in order to achieve the above object, there is provided a program storage means for storing a program for processing an input to control a musical instrument and a program for generating musical tones, and the program. address control circuit means for controlling the address of the storage means; data storage means for storing data necessary for generating musical tones; arithmetic circuit means for performing various operations; An electronic musical instrument comprising a microcomputer comprising an operation control circuit means for controlling the operation of each means, the microcomputer generating musical tones by executing a program for generating musical tones in the program storage means. A processing device is provided.

As described above, in the present invention, there is an immeasurable advantage of the electronic musical instrument processing device of this new architecture, which does not require any sound source circuit hardware for generating musical tones. The first is the degree of freedom in design. In other words, changes in the number of polyphonic sounds per sound and changes in the tone synthesis method can be handled by changing the design of the program.

Second, since no sound source circuit hardware is required, the overall circuit scale can be significantly reduced.

In the conventional case, since the circuit scale of the sound source circuit LSI chip is large, there is a limit to the yield in chip manufacturing (the yield is almost inversely proportional to the chip area).

Therefore, the present invention can significantly reduce the manufacturing cost of electronic musical instruments.

In one configuration example, the program storage means comprises a read-only memory (1'lOM). The microcomputer is realized by an integrated circuit chip, and in addition to the above-mentioned means, this chip also includes a digital-to-analog (D/A) converter for converting the generated digital musical tone signal into an analog signal, and a port for receiving input for controlling the musical instrument. Implemented. Also,
A multiplication circuit used for calculating waveform data is also added to the calculation circuit means.

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

The overall configuration of the electronic musical instrument according to this embodiment is shown in FIG. 1, and the entire device is controlled by a microcomputer l. In particular, according to the present invention, not only the processing of the control input of the I! A switch unit 4 consisting of a keyboard 2 and an atrs key 3 is a control input source for the musical instrument, and information input from the switch unit 4 is processed by a microcomputer l. The analog-converted musical tone signal generated by the microcomputer 1 is filtered by a low-pass filter 5,
The sound is amplified by an amplifier 6 and emitted through a speaker 7.

The IT source circuit 8 supplies the necessary power to the microcomputer 1, the low-pass filter 5, and the amplifier 6.

The internal structure of the microcomputer I is shown in a block diagram in FIG. Each of the illustrated elements is mounted on a single chip. The actually manufactured chip has a chip size of 5 x 5 mm, has an eight-tone polyphonic simultaneous sound generation, and uses a PCM (waveform readout method) as the musical tone synthesis method, but the present invention can be applied to other polyphonic number 2 and other musical tone synthesis methods. can also be applied.

The control ROM 31 stores programs for processing various control inputs of the musical instrument and programs for generating musical tones, and a program M (command) at a specified address is sent from the ROM address control unit 39 via the ROM address decoder 32. Note that in the specific embodiment, the program word length is 28 bits, and a part of the program word is sent to the ROM address control unit 39 as the lower part (in-page address) of the address to be read next. Although the system uses a netast address input method, the present invention can also be applied to a program counter system. The RAM address control unit 33 specifies the address of a corresponding register in the RAM 34 when an operand of an instruction from the control ROM 31 specifies a register. The RAM 34 is a group of registers and is used for general-purpose operations, flag operations, musical tone operations, and the like. Adder/subtractor and logic operation unit 35 and multiplier 3
6 is used when the instruction from the control ROM 32 is an arithmetic instruction. In particular, the multiplier 36 is used to calculate musical waveforms, and as an optimization for this purpose, it multiplies the first and second data inputs (for example, 16 beat 7 bit data) to obtain data of the same length (16 bits) as the input. It is designed to output . The RAM 34, adder/subtracter 351 and multiplier 36 allow
An arithmetic circuit (AU) is configured. The control data reed waveform ROM 37 contains various musical tone control parameters such as pitch data and envelope data (rate, level), and a PC.
M (pulse code modulation) tone waveform data is stored. Envelope data and musical waveform data are prepared for each musical tone.

Operation skin analysis section (operation control circuit) 38
decodes the operation code of the instruction from the control ROM 31,
The analyzer sends control signals to each part of the circuit in order to perform the instructed operation.

In order to execute the musical tone generation program in the control ROM 31 at predetermined intervals, this embodiment employs timer interaction. That is, the ROM address control unit 39 sends a control signal (interrupt request signal) at regular intervals by the interwoven control unit 40 having a timer, and in response to this signal, the ROM address rFAW unit 39 executes the command of the next main program. Save (retain) the address of
The start address of the interwoven processing program (subroutine) in which musical tones are generated is set instead. This starts the interwoven processing program. Since there is a return instruction at the end of the interwoven processing program, this return instruction is used by the operation analysis unit 38.
When the address is decoded, the ROM7 address control unit 39 sets the saved address again and returns to the main program.

The manual port 41 and output port 42 are #! 12. Used for key scanning of function key 3. The musical tones generated in the interwoven processing program are converted into analog signals by the digital/analog converter 43 and output to the outside.

FIG. 3(A) shows the flow of the main program of the microcomputer 1 of this embodiment. AI is the initial processing when the power is turned on, and the RAM of the microcomputer 1
Clears (register group) 34, sets initial values such as rhythm tempo, etc.

At A2, the microcomputer 1 outputs a signal for key scanning from the output port 42, and takes in the state of the switch section 4 from the input port 41, so that the function key, tI
The state of the IIi key is stored in the keeper 7 area of the RAM 34. In A3, from the state of function key 3 obtained in A2,
Identify the function key whose state has changed and execute the instructed function (for example, set musical tone number, set envelope number, set rhythm number, etc.). In A4, change from the state 1IIi2 obtained in A2. I did it! (Key press, key release) is identified In the zero-order A5, from the processing result of A4, key assignment processing for sound generation processing A9 is performed.In A6, when the demo performance key is pressed with function key 3, control data The sound generation process A is performed by sequentially reading and processing the demo performance data (sequencer data) from the dual waveform ROM 37.
Performs key assignment processing etc. for A7, and when the rhythm start key is pressed, the control data and waveform RO
Sequentially read rhythm data from M37 and perform sweat processing A9
In step 8, in order to know the timing of events required in the main flow, we perform key assignment processing for the flow-period timer processing. It is obtained by pre-counting the number of rhabdos. This counting process is performed in the interwoven timer process B3 described later.), and the envelope river timer (envelope calculation period) and reference value for rhythm are calculated. Get 1 pronunciation processing A9 in A5,
A6. From the data set in A7, various calculations are performed to actually generate musical tones, and the results are set in the sound source processing register (sixth vli) in the RAM 34. AIO
is a preparation process for the next main flow pass, and NEW ON indicates the change to the key press state obtained in this bus.
NEW indicating that the status is ON or the key is released
Processing such as changing the OFF state to OFF is performed.

Figure 3B shows the flow of the interwoven processing program in which musical tones are generated.In B1, musical sound waveform data (cumulative waveform values for 8 tones) generated in the previous interwoven sound source processing B2 is transferred to a D/A converter. Send to 43. As a result, the zero-order sound source processing B2, in which musical tone samples are given to the D/A converter 43 at a constant cycle, is the key point of the embodiment, and conventionally this processing was performed on the sound source circuit hardware. . In the zero-order interwoven timer process B3, which will be described in detail later, the timer register for flow-cycle timing (
Add 1 each time you pass (within 11AM34).

Note that in this embodiment, the registers to be written by the main program in the interwoven processing program are as follows:
Since the contents are not rewritten, the register saving and restoring processing that is normally performed at the start and end of interwoven processing is unnecessary. In other words, since the registers in the RAM 34 related to musical tone processing and the registers related to other processing are independent, the transition from the main program to the interwoven processing is performed with as little delay as possible.

The details of sound source processing B2 are shown in Figure 3C. After clearing the waveform addition RAM area (R6 diagram) in CI, processing 02 to c9 for 8 channels is performed in order. Finally, the tone waveform value of the channel is added to the data in the waveform addition RAM area.

Figure 4 shows the flow of the operation of the embodiment over a period of 5 hours, and shows the flow of the operation of the example L6. A, B, C, D, E, and F are fragments of the main program (FIG. 3A), and interwoven processing (FIG. 3B) is executed at regular intervals. The time chart is as shown in FIG. 5. As shown in FIG. 5, each time an interleaved state is entered, a musical waveform signal is transferred to the D/A converter 43, and a corresponding analog signal is output to the outside.

FIG. 7 shows in detail the processing from 02 to C9 in FIG. 3C for one channel. Channel processing can be broadly divided into envelope processing (D1 to D7) and waveform processing (
08-021).

FIG. 8 shows an envelope generated by envelope processing. The envelope of one musical note consists of several steps (segments).
It is shown in segments.

ΔX in the figure is the sampling period of the envelope,
Δy is the change width of the envelope value.

Channel envelope processing IM (DI-07) -
c performs envelope processing calculations for each sampling time and checks whether the target level of the step has been reached. When they match, the target value is set in the current envelope register (see Figure 6), so this is detected in the main program's sound generation process A9 and the data (ΔX, Δy, target value) for the next step's envelope is set. envelope value) is set in each register.

In detail, at 01, the timer register for comparing with the envelope calculation cycle ΔX is incremented for each interconnect, and when it matches ΔX at D2, the addition/subtraction flag (sign bit) of the envelope displacement data Δy is tested at D3. to determine whether the envelope is rising or falling, D4 and D5 subtract or add the current envelope, respectively.D6 checks whether the current envelope has reached the target envelope value, and if it has, Set the target level to the current envelope. As a result, data for the next envelope step will be set in the sound generation process A9 of the main program. Also, pronunciation processing A9-1? When a current envelope of zero is read, it is treated as the end of pronunciation.

Next, waveform processing D8 to D21 will be described.

In waveform processing, the sa part of the current address is used to process the waveform RO.
Read the waveform data of two adjacent addresses from M,
The expected waveform value for the current address indicated by (integer part to decimal part) is obtained by interpolation. The reason why interpolation is necessary is that the waveform sampling period by Interabdo is constant, and the address addition value (pitch data) spans a certain range in application to musical instruments. If you prepare waveform data, there is no need for interpolation, but it will result in an unacceptable increase in storage capacity).Tone deterioration and distortion due to interpolation are more pronounced in the high range, so
Normally, the original sound is reproduced at a faster cycle than the recording sampling cycle of the original sound. In this embodiment, the reproduction cycle of the original sound (A4) is doubled (FIG. 9). Therefore, when the address addition value is 0.5, the A4 sound can be obtained. In this case, the address addition value for A#4 is 0.5
29, and when A3, it becomes l. These address addition values are stored as pitch data in 8m data/waveform ROM3.
7, and when a key is pressed, in the sound generation process A9, the pitch data corresponding to the key, the waveform start address, waveform end address, and waveform loop address of the selected tone are stored in the corresponding register of the RAM 34, that is, address addition Set in the value register, start address/current address register, end address register, and loop address register.

For reference, in FIG. 10, which shows interpolated waveform data with respect to time, white circles indicate waveform data values at addresses in the waveform ROM, and black circles indicate interpolated values.

There are various methods of supplementary rules, but here we use linear supplement + 1JI
is adopted. Waveform raw dt flow D8 to D21 in Fig. 7
To describe this in detail, first, in D8, an address addition value is added to the current address to obtain a new current address. D9 compares the current address and the end address, and if the current address > the end address, 010.011 is used, and if the current address is less than the end address, 012 is used to specify the next physical (address) or logical (operational) address. The waveform ROM is accessed using the integer part at 014 to obtain the next waveform data. The loop address is operationally the next address after the end address. That is, the ninth
In the case of the figure, the illustrated waveform is read out repeatedly. Therefore, when the current address = end address, the waveform data of the loop address is read as the next address (D
13).

D15. D16 causes the waveform R to be generated at the integer part of the current address.
Access the OM and read the current waveform data. Next, subtract the current waveform value from the next waveform value in D17, and
By multiplying the difference by the decimal part of the current address at 019 and adding the result to the current waveform value at 019, the linearly interpolated value of the waveform is obtained. This linearly interpolated data is multiplied by the current envelope value to obtain the musical tone data value of the channel m(D
20), and adds it to the contents of the waveform addition register to accumulate musical tone data (021).

Finally, to describe the circuit scale and operating time of a specific example (8-note polyphonic PCM sound source system), the control ROM is 112 Kbits, and the RAM 34 is 5.4 Kbits.
i t, control data/waveform ROM37 (tone is 10
0 tone) is 508 Kbits. One machine cycle is about 276 nanoseconds, and the maximum number of cycles of an interwoven processing program during operation is about 150. The execution interval (musical tone output sampling period) of the interwoven processing is approximately 47 microseconds.

As described above, in the embodiment, the microcomputer 1 generates musical tones in the timer-interrupted processing program, so conventional tone generator circuit hardware is not required, reducing the circuit scale and improving yield. This can lead to improved performance, reduced costs, and a high degree of freedom in design.

[Modification J This concludes the description of the embodiment, but various modifications and changes are possible without departing from the scope of the present invention.

For example, in the above embodiment, musical tones are generated by applying a timer interlude and executing an included program in order to output musical waveform samples at predetermined time intervals.

By incorporating the P command) into the program.

Processing instead of the included processing described above (hereinafter referred to as fixed time processing) may be executed at fixed time intervals (see Figure 11). In other words, the execution time of each instruction of the program is determined by the master clock. Since it can be determined, a fixed time processing program for generating musical tones can be inserted as a subroutine between the main program for a fixed amount of time.

In order to secure the fixed time, all branches in the fixed time processing program of the main program and subroutine must be processed in the same time.

For example, assume that there is a flow as shown in FIG. 12 in the main program, and that processing is performed for a certain period of time before and after the flow. For simplicity, assume that a branch instruction takes 2 hours and 2 normal instructions takes 1 time. As shown in FIG. 12, the time from the first branch to the fixed time processing is 2. 5 when taking route B, 6 when taking route C. When passing through route d, the value is 5, which varies depending on the route. Route) If four dummy instructions are placed in route a, one dummy instruction is placed in route b, and one dummy instruction is placed in route d, the times of all routes are unified to 6 (Figure 13).

Even within the fixed time process, if the processing time varies depending on the branch, the time until the next jump to the fixed time process will not be constant. Therefore, it is necessary to insert a dummy instruction to make the processing time required for all branch routes constant even within the fixed time processing. - As an example, FIG. 14 shows a case in which a dummy command is inserted into the musical tone generation channel processing within the fixed time processing.

In the embodiment and the above-mentioned modification, the sampling period for musical tone generation is always kept constant. It is also possible to generate musical tones with a variable sampling period.0 For example, in the PCM sound source method, instead of performing waveform interpolation processing, the waveform data at each address in the waveform ROM is used as is, and the time that advances by l addresses is used. (the time from reading the waveform data of a certain address to reading the waveform data of the next address) is proportional to the frequency a (pitch data) of the musical tone 1 For example, for A4, time l, for A3, 2 The time shall be . The control of this time interval is 1 example.

This can be achieved by presetting data corresponding to the frequency of musical tones in the internal timer of the timer interworking section w and performing interworking every time the timer runs out. However, in the case of polyphonic application, the interval between interwovens changes depending on the frequency of the musical tone used in each channel, so a timer interwoven control section and a timer interwoven processing program are provided for each channel. Also, D/A
A converter is also provided for each channel, and its conversion period is made to match the period of timer interwoven processing of the corresponding channel.

In all of the above examples, the musical tone generation operation in the microcomputer and the operation of the D/A converter are basically in synchronization. Instead of this, an asynchronous method can also be considered.
For example, the musical tone generation output of a microcomputer and the D/A
A buffer mechanism is provided between the converter and the converter. To give a microcomputer the power to generate a sample sequence of musical tones at a higher speed than the conversion period of a D/Af# device. This can be achieved, for example, by distributing a large number of subroutines for generating musical tones in the main program. The sample sequence of musical tones from the microcomputer is written into a buffer, and the D/A
It is read out in synchronization with the operation of the converter and input to the D/A converter. The time from one musical sound generation subroutine to the next subroutine varies slightly depending on the execution state of the program, but assuming it is shorter than the conversion cycle of the D/A converter, writing a musical sound sample string to the buffer will fill the remaining buffer. As the capacity decreases, the average calculation cycle of the musical tone sample sequence of the microcomputer is changed according to the remaining capacity of the buffer. In other words, a branch instruction is provided in the musical tone generation subroutine so that musical tone generation is performed selectively in the subroutine depending on the remaining capacity of the buffer. The musical tone is generated twice. In one configuration example, a buffer mechanism and a D/A converter are prepared for each sound generation channel, and the conversion speed of the D/A converter is configured to be variable. This is because the ratio between the phase difference between the samples of the musical sound sample sequence generated by the microcomputer and the frequency of the musical sound maintains high quality sound.
This can be used when switching in a limited range (for example, an octave range).The important thing is that the output of the D/A converter will not match the intended musical tone, regardless of the phase difference between the sample sequences. It has a frequency. The microcomputer shall maintain the ratio between the phase difference between samples and the frequency of the musical tone within the selected range (for example, within the range of ! otaves, the microcomputer shall maintain the ratio of the phase difference between musical samples to the frequency of the musical tone). In this case, the microcomputer stores data corresponding to the above ratio in D/
It is applied to a frequency control circuit (which can be configured with a programmable timer or the like) that controls the conversion cycle of the A converter (and therefore also the read cycle of the buffer). This circuit generates control timing signals for eight-way reading and D/A conversion according to the received ratio data. As a result, D
The output of the /A converter becomes an analog signal having the frequency of the musical tone.

For example, if the phase difference between musical tone samples generated by a microcomputer is 17N, which corresponds to one period of the musical tone, and the period of the musical tone is P, the D/A converter will generate one signal in a time of P/N. Convert digital musical sound samples to analog signals. Therefore, the time required to convert N digital tone samples is equal to the period P of the tone, and an analog signal having the frequency (1/P) of the tone is obtained.

In addition, various variations are excretable.

[Effects of the Invention] As described above, according to the present invention, when a microcomputer executes a program for generating musical tones,
Since musical tones are generated, there is no need for conventional sound source circuit hardware. Therefore, the circuit scale of the entire device can be reduced, yield can be increased, and costs can be reduced. Furthermore, the degree of freedom in design is increased, and changes in musical tone synthesis methods and polyphonic numbers can be easily accommodated.For example, in the conventional technology, a four-tone polyphonic sound source system and a PCM sound source system were realized with a 7x7mm chip. However, in the embodiment of the present invention, an 8-tone polyphonic
We were able to realize the PCM sound source system (including a D/A converter) with a 5x5mm chip. Generally, it is considered that the circuit scale can be reduced by half or less.

[Brief explanation of the drawing]

FIG. 1 is an overall configuration diagram of an electronic entertainment device according to an embodiment of the present invention, FIG. 2 is a block diagram showing the configuration of a microcomputer according to the embodiment, and FIG. 3A is a diagram showing the flow of the main program of the microcomputer. Figure 3B is a flowchart of the interwoven processing in which musical tones are generated, Figure 3C is a detailed flowchart of the sound source processing in Figure 3B, Figure 4 is a diagram showing the flow of the program over time, and Figure 5 is a diagram showing the flow of the program over time. 6 is a diagram showing a table of the musical tone generation RAM placed in the RAM 34 of FIG. 2, and FIG. 7 is a detailed flowchart of one channel processing of FIG. 3C. , FIG. 8 is a diagram showing an envelope, FIG. 9 is a diagram showing waveform data of the waveform ROM, FIG. 10 is a diagram showing an interpolated calculation waveform along time, and FIG. Figure 12 is a diagram that formally shows a part of the main flow, and Figure 13 is a modification of the flow in Figure 12 for constant time processing. FIG. 14 is a diagram showing a modified flow of channel processing for fixed time processing, and FIG. 15 is a block diagram showing an example of conventional sound source circuit hardware. l...Microcomputer, 31...
・Control ROM, 34...RAM, 35...
. . . Adder/subtractor and logic operation unit, 36 . . . Multiplier, 37 . . . Control data/waveform ROM, 38.
...Operation analysis section, 40...Interfaced control section, 41.Old...Input port, 42..
...Output port, 43...D/A converter. Patent applicant Casio Computer Co., Ltd. Figure 3A May'/70- Procedural amendment tooth (method) May 24, 1999

Claims (4)

[Claims] (1) Program storage means for storing a program for processing input to control a musical instrument and a program for generating musical tones, and address control circuit means for controlling the address of the program storage means, necessary for generating musical tones. a data storage means for storing data; an arithmetic processing circuit means for decoding each instruction of a program in the program storage means to control operations of the address control circuit means, the data storage means, and the arithmetic processing circuit means; 1. A processing device for an electronic musical instrument, comprising a microcomputer comprising: an operation control circuit means; and the microcomputer generates musical tones by executing a program for generating musical tones in the program storage means. (2) A processing device for an electronic musical instrument according to claim 1, wherein the program storage means is constituted by a read-only memory. (3) The processing device for an electronic musical instrument according to claim 1, wherein the microcomputer is composed of an integrated circuit chip;
A processing device for an electronic musical instrument, characterized in that the chip is further provided with a digital-to-analog converter for converting a digital tone signal into an analog signal, and a port for receiving input for controlling the musical instrument. (4) A processing device for an electronic musical instrument according to claim 1, wherein the arithmetic circuit means includes a multiplication circuit for calculating waveform data.