JPH02179695A - Processor for electronic musical instrument - Google Patents

Abstract

PURPOSE:To improve the quality of musical sounds by providing a latch controlled by a program control signal and providing another latch, into which the output signal of a latch is taken at the timing of an accurate sampling period signal, between the output of the latch and the input of a D/A converter. CONSTITUTION:An interrupt control latch 46 controlled by the accurate timing signal from an interrupt control part 40 is provided between a soft control latch 45 controlled by the program control signal from an operation analyzing part 38 ad a D/A converter 43A which converts a digital musical sound signal to an analog musical sound signal. The generation period of the interrupt signal is very stable because according with the stability of a clock oscillator. Since the output of the latch 46 is switched synchronously with the interrupt signal, switching of input data of the converter 43A is synchronized with the interrupt signal. Consequently, the conversion period in the D/A conversion time is stably kept, and distortionless musical sounds 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 a technique for converting a digital musical tone signal generated under program control of a microcomputer into an analog signal.

[Background of the invention] 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 instrument (from the i and the console panel). inputs, MIDI and other external control inputs, inputs from internal or external performance memories, etc.) and transfers appropriate commands to the tone generator circuit.

There are several problems with the system architecture of electronic musical instruments, in which musical tone generation processing is performed by sound source circuit hardware, and musical instrument control input processing is performed by a microcomputer. First, the tone generator circuit hardware consists of storage devices that temporarily store data and arithmetic circuits that perform calculations at various stages of processing musical tone parameters, so the circuit size inevitably increases. Typically, it is about twice the size of a microcomputer. Second, there is a limit to the ability of a microcomputer to variably control the way musical tones are synthesized using sound source circuit hardware.For example, the number of simultaneous polyphony is fixed in hardware, so a microcomputer It cannot be changed using commands from .

This control limit becomes a barrier when designing new sound source circuit hardware. That is, large-scale circuit changes are often necessary, which requires a great deal of development time and effort.

Furthermore, the communication protocol or interface (transfer method,
Command sets, etc.) also need to be re-examined and re-developed for each sound source circuit hardware.

For the above reasons, the applicant has been researching the system architecture of a new electronic musical instrument processing device that can generate musical tones solely through microcomputer program control without using sound source circuit hardware. As a result, I saw it come true.

However, in a structure where the microcomputer itself generates musical tones, due to the nature of program control, it is impossible to keep the period of the sample sequence of the digital musical tones sent from the microcomputer to the digital analog CD/A) converter completely constant. Possible or very difficult. That is, since the amount of work (processing amount) to be processed by the microcomputer changes from time to time due to inputs to the microcomputer, etc., the amount of processing for generating musical sounds included in the processing target also changes. This essentially means a fluctuation in the generation cycle of digital musical tones. When a digital musical tone that changes at an unstable period is converted into an analog signal, distortion occurs in the musical tone, which poses a major problem for electronic musical instruments.

[Object of the Invention] Therefore, the object of the present invention is to provide a processing device for an electronic musical instrument that can extract digital musical tones generated by a microcomputer itself at an accurate sampling period and output them as an analog signal with less distortion. be.

[Structure and operation of the invention] According to the present invention, in order to achieve the above object, in an electronic music 2 processing device in which a microcomputer itself generates musical tones under program control, a digital musical tone signal generated by the microcomputer is processed. A first latch means that latches at the timing of a program M control signal from a microcomputer, and an output of this first latch means and an input of a digital-to-analog converter (usually a control gate of a switch called a bit switch). and a second latch means provided between the second latch means and the second latch means for latching the output signal from the first latch means at the timing of an accurate sampling period signal.

In the case of this aIO1t4, digital/analog (D/A
) The digital musical tone signal supplied to the input of the converter means is
Due to the action of the second latch means, switching occurs at accurate timing of the sampling period signal. This means that the digital to analog conversion period in the digital-to-analog converter means can be determined by anyone with the accuracy of the sampling period signal. Therefore, distortion caused in the process of converting a digital musical tone signal to an analog musical tone signal is minimized, and a high quality acoustic signal can be output to the outside.

- In the configuration example, the microcomputer is realized by an integrated circuit chip, and on this chip - E-recording means are added, and a digital-to-analog (D/A) converter for converting the generated digital musical tone signal into an analog signal, and a 2J ! A port that receives input to control the bases is also implemented.

[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. That is, the microcomputer 1 executes not only the processing of control inputs for the musical instrument but also the processing of generating musical tones under program control, and does not require any sound source circuit hardware for generating musical tones. The switch section 4 is a control input source for the musical instrument, and information input from the switch section 4 is processed by the microcomputer 1. The digital musical tone signal generated by the microcomputer l is converted into an analog signal by a digital-to-analog converter (located in block l in FIG. 1), then filtered by a low-pass filter 5, amplified by an amplifier 6, and then sent to a speaker 7. The sound is emitted through. A power supply circuit 8 supplies necessary power to the microcomputer 1, low-pass filter 5, and amplifier 6.

The internal structure of the microcomputer 1 is shown in a block diagram in FIG. 2. Each element shown in FIG. 0 is mounted on one chip. The one we actually produced had a chip size of 5 x 5 mm and an 8-note boliphony.

Although the musical tone synthesis method is PCM (waveform reading method), the present invention can be applied to other polyphonic numbers and other musical tone synthesis methods.

The control ROM 31 stores a program for processing various control inputs of the musical instrument and a program for generating musical tones, and a program word (instruction) at a specified address is sent from the ROM address control unit 39 via the ROM address decoder 32. 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 (intra-page address) of the address to be read next. Although the input netast address method is used, a program counter method may be used instead.

When the operand of the instruction from the control ROM 31 specifies a register, the RAM address control unit 33
Specifies the address of the corresponding register in M34. R
AM34 is a register group, and is used for general purpose operations, flag operations,
Used for calculations of musical tones, etc. Adder/subtractor and logic operation section 3
5 and the multiplier 36 are used when the instruction from the M operational ROM 32 is an operation instruction. In particular, the multiplier 36 is used to calculate the musical sound waveform, and the first and second multipliers are optimized for this purpose.
The data input (for example, 16 bit data) is multiplied and data of the same length (16 bits) as the input is output. The above RAM 34, adder/subtractor 35, multiplier 3
6 constitutes an arithmetic circuit (AU). The control data/waveform ROM 37 stores various tone control parameters such as pitch data and envelope data (rate, level), and PCM (pulse code modulation) tone waveform data. Envelope data and tone waveform data are prepared for each tone of tone a.

Operation analysis section (operation control circuit) 38
decodes the operation code of the instruction from the control ROM 31,
Control signals are sent to each part of a circuit to carry out 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 interwoven control unit 40 having a timer (hardware counter) sends a control signal (interrupt request signal) to the ROM address control unit 39 at regular intervals, and this signal causes the ROM address control unit 39 to select the next row The address of the instruction of the main program is saved (held), and the start address of the interwoven processing program (subroutine) in which musical tones are generated is set instead. This starts the interact processing program. Since there is a return instruction at the end of the interwoven processing program, when this return instruction is decoded by the operation analysis section 38, the ROM address control section 39 again writes the saved address up - 2, and the main program to return to. Note that the interconnected control unit 40 is the microcomputer 1 (C
Although it is depicted as an internal element of the microcomputer (PU), it requests the microcomputer to stop its current work and perform special processing, and logically it is an external element (peripheral device) of the microcomputer. be.

Input port 41 and output port 42 are the key! 2.41 Function Used for key scanning of 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 imports the state of the switch section 4 from the input port 41, thereby scanning the function key 1.
The state of the 191 keys is stored in the key buffer area of the RAM 34. In A3, the function key whose state has changed is identified from Ja1# obtained in A2, the new state of key 3, and the previous state, and the instructed function is executed (for example, setting the musical note number, setting the envelope number, etc.). set, rhythm number set, etc.), A4 identifies the key that has changed (key press, key release) from the latest state and previous state of the key 912 obtained in A2. Processing result of A4 in 0-order A5. From, 3? ! Sound processing A
In A6, when the demo performance key is pressed with the 7 key 3, the demo performance data (sequencer data) is sequentially read from the control data/waveform ROM 37 and processed. 5 Pronunciation processing A9
In A7, when the rhythm start key is pressed, the control data and waveform ROM is
The rhythm data is sequentially read from 37 and the key assignment processing for 1 sound processing A9 is performed.In 8, flow-period type processing is performed.In order to know the timing of events required in the main flow, flow-same time (this is It is obtained by counting the number of timer interrelated operations executed during one cycle of 70-.This counting process is performed in the interrelated timer processing B3 described later.) yI calculation is performed based on (Envelope calculation cycle) and rhythm reference value A9 0 sound generation processing 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 (FIG. 6) in the RAM 34. A10 is a preparation process for the next main flow bus, which includes changing the NEW ON state that indicates a change to the key pressed state obtained in this pass to ON, or NEW indicating a change to the key released state.
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, the musical sound waveform data (accumulated waveform value for 8 tones) generated in the previous interwoven sound source processing B2 is D/A converted. In zero-order sound source processing B2, which is sent to the processor 43, musical sound waveform data for each channel is generated, accumulated, and stored. Conventionally, this process was performed by the sound source circuit hardware, but in the zero-order interwoven timer process B3, the timer register (R
Add 1 each time you pass AM34).

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, there is no need to save and restore registers, which is normally performed at the start and end of interwoven processing.

Details of sound source processing B2 are shown in Figure 3C. After clearing the waveform addition RAM area (see Figure 6) in which accumulated waveform values are stored in the CI, processes C2 to C9 for 8 channels are performed in order. ing. At the end of each channel process, the tone waveform value of the channel is added to the data in the waveform addition RAM area.

FIG. 4 depicts the flow of the program control operation of the embodiment. A, B, C, D, E, and F are fragments of the main program (Fig. 3A), and are the fragments of the interactive control section 40.
Interrupt processing (FIG. 3B) is executed every time there is an interrupt request from . The flow of this operation is shown in a time chart as shown in FIG. In FIG. 5, the interwoven signal generation interval T is very stable. because,
This is because the interwoven signal is generated by the hardware counter of the interwoven control section 40.

Therefore, although not shown, its stability is determined by the stability of a clock oscillator (typically a crystal oscillation or the like). In this embodiment, the main processing is interrupted by this interwoven signal, and musical tone generation processing (interwoven processing) is performed, thereby making the generation sampling period of musical tones constant. Indeed, this approach makes it possible to make the average period of musical tone generation sampling equal to the generation interval T of the interwoven signal.

Nevertheless, as highlighted in FIG. 5, the timing at which interwoven processing actually begins may vary. This variation is due to program control. That is, even if the microcomputer 1 is interrupted from the outside, it is impossible for the microcomputer 1 to immediately interrupt the operation being executed, so the microcomputer 1 enters the interactive processing after the execution is completed. Furthermore, while a process is in a process that is undesirable to be interrupted, interrupts may be masked to prevent inter- rupted processing from occurring until a series of operations for that process is completed. Since the transition to interwoven processing depends on the process being executed at that time, the period of musical tone generation becomes unstable. Specifically, the timing of the process executed in step Bl in FIG. 3B, that is, the process of taking out the digital musical tone data in the waveform addition register in the RAM 34 and setting it in the port of the D/A converter 43, is different. It shifts to If the sampling period of the D/A converter 43 is the same as the execution interval of step Bl, significant distortion will occur in the process of converting from digital to analog. This invention solves this problem.

The means will be detailed later.

Next, channel processing will be explained. 0 in Figure 3C
FIG. 7 shows details of the processing of steps 2 to C9 for one channel. Channel processing can be broadly divided into envelope processing (DI-07) and waveform processing ('D8 to D2).
Consists of 1).

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 % [(Dx to D7) calculates the envelope 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 processing A9 and the data for the next step's envelope (ΔX, Δy, target envelope value) is set in each register.

To explain in detail, yi calculation period ΔX of the envelope is 01
Increment the timer register for comparison with ΔX for each interconnect, and when it matches ΔX in D2, test the addition/subtraction flag (sign bit) of the envelope displacement data Δy in D3 to determine whether the envelope is rising or falling. Then, in D4 and D5, the current envelope is subtracted or added, respectively.In D6, it is checked whether the current envelope has reached the target envelope value, and if it has, the target level is set in 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. Furthermore, when the current envelope of zero is read by the sound generation processor JIA9, it is processed as the end of sound generation.

Next, waveform processing D8-021 will be described.

In waveform processing, the integer part of the current address is used to convert 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 calculated using supplementary rules. 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 the waveform data, there is no need for supplementary rules, but it will increase the storage capacity which is unacceptable!), as the deterioration and distortion of the timbre due to interpolation are more pronounced in the high range,
Normally, the original sound is reproduced at a faster cycle than the recording sampling cycle of the original sound. In this embodiment, the period of reproduction 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 the control data/waveform ROM 3.
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 It is set in the value register, start address blue, current address register, end address register, and loop address register.

For reference, in FIG. 9 showing interpolated waveform data with respect to time in FIG. 10, white circles indicate waveform data values at addresses in the waveform ROM, and black circles indicate interpolated values.

There are various interpolation methods, but here we use linear interpolation. Describing the waveform generation processing D8-021 in FIG. 7 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 end address, and if current address>end address, 010. When the current address is the end address, Dll calculates the next physical (address h) or logical L (operational) address using D12.
At 014, the waveform ROM is accessed using the integer part to obtain the next waveform data. The loop address is operationally the next address after the end address. That is, in the case of FIG. 9, the illustrated waveform is repeatedly read out. Therefore, when 2 current address = end address, read the waveform data of the loop address as the next address (D13)
.

By D15 and DIB, waveform R is 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 (D
20), add it to the contents of the waveform addition register and accumulate the musical tone data (021), the digital musical tone data for all channels accumulated in this register will be D/D in step B1 at the next interrupt. The signal is sent to the A converter 43.

As mentioned in connection with FIG. 5, the sampling period for musical tone generation by the microcomputer l is not strictly constant; FIG.
This configuration is set to the conversion period of the A converter 43. That is,
Soft control latch 45 as a port of D/A converter 43
This latch 45 is controlled by a Pumgram control signal from the operation analysis unit 38, and the output of the latch 45 is connected to a corresponding bit switch (not shown, typically a current control type electronic switch) in the block 43A. ) to the control gate. Block 43A is where the digital signal is converted into an analog signal, so below, D
/A converter. That is, Fig. 11 (A
), step Bl of the interwoven processing program
, under the control of the operation analysis unit 38.

The waveform addition register in RAM 34 is specified.

The latest digital musical tone data stored therein is retrieved and put on the data bus, and at the timing when the digital musical tone data is on the data bus, a strobe program control signal is sent from the operation analysis section 38 to the latch 45's gate input. The data on the data bus is set, and new digital musical tone data is sent from the latch 45 to the D/D.
The signal is input to the A converter 43A. Therefore, Fig. 12 (
As shown in A), the digital musical tone data input to the D/A converter 43A is switched at an unstable cycle due to program control. If the conversion period (sampling period) of the D/A converter 43A is very stable,
For example, if the machine cycle of a microcomputer I is several tens or hundreds of nanoseconds, even this one machine cycle delay causes the D/A converter 3143A to reach the audible frequency. The accuracy of the conversion cycle required to faithfully convert a digital signal into an analog signal is too high. In other words, even a difference on the order of nanoseconds causes distortion that is perceptible to human hearing.

This problem can be solved by adopting a configuration as shown in FIG. 11(B). In other words, the operation analysis section 3
An interlaced signal, which is a precise timing signal from the interlaced control section 40, is connected between the soft control latch 45 controlled by the program control signal from the interconnected controller 40 and the D/A converter 43A that converts the digital musical tone signal to an analog musical tone signal. A signal controlled interlaced control latch 46 is provided. The generation period of the interwoven signal follows the stability of the clock oscillator, so it is extremely stable. The output of the latch 46 is switched in synchronization with the timing of the interlaced signal. That is, the generation period of the interwoven signal becomes the conversion (sampling) period of the D/A converter 43A. 11th
Figure 12 (B) shows a time chart for the configuration shown in Figure (B).
), the timing at which the output of the latch 45 is switched varies depending on the timing shift of the interwoven processing, but the latch 46 operates on the interwoven signal.
Therefore, the timing at which the input data of the D/A converter 43A is switched is synchronized with the interwoven signal. latch 4
6, the digital musical tone signal input to the D/A converter 543A is delayed by one period of the interwoven signal on average, but this delay is not a problem at all.For example, the period of the interwoven signal is 47 microseconds. can be.

Such a short delay cannot be detected by human hearing (usually a few milliseconds is the perceptible limit).

This concludes the description of the embodiment, but various modifications are possible without departing from the scope of the invention. Although this is done by executing a subroutine, it may also be done by a subroutine that does not depend on an interrupt. In that case, the node Operation instructions (NOP instructions, dummy instructions) can be distributed in the program, and any method and polyphonic number can be used as far as the microcomputer's capabilities allow for the musical tone synthesis method and polyphonic number to be executed in the program. obtain. In short, the present invention is applicable to all configurations in which a microcomputer itself generates musical tones under program control.

[Effects of the Invention] As described above, in this invention, in an electronic music evaluation processing device configured such that a microcomputer itself generates musical tones under program control, the digital musical tone signals generated by the microcomputer are controlled by the program control from the microcomputer. a first latch means for capturing by the signal, inserted between the output of the first latch means and the input of the digital-to-analog converter means, for augmenting the output signal from the W4i latch means with a precise sampling period signal; Since the second latch means is used, the conversion period in the digital-to-analog converter means can be kept stable, and musical tones with less distortion can be output to the outside.

[Brief explanation of the drawing]

FIG. 1 is an overall configuration diagram of an electronic musical instrument according to an embodiment of the present invention, FIG. 2 is a block diagram showing the configuration of a microcomputer according to the embodiment, FIG. 3A is a diagram showing the flow of the main program of the microcomputer, and FIG. 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 of Figure 3B, Figure 4 is a diagram showing the flow of the program along the verses, and Figure 5 is the 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 shows the envelope, Fig. 9 shows the waveform data of the waveform ROM, Fig. 1O shows the interpolation calculation waveform along time, and Fig. 11 shows the sampling period of the D/A converter. A diagram comparing a configuration that becomes unstable and a configuration in which the sampling period of the D/A converter is stabilized according to the present invention. Figure 12 shows the time when the sampling period of the D/A converter is unstable. It is a figure which compares and shows a chart and a time chart in a stable case. l...Microcomputer, 31...
・Control ROM, 34...RAM, 35...
. . . Adder/subtractor and logical filter unit, 36 . . . Multiplier, 37 . . . Control data/waveform ROM, 38.
...Operation analysis section, 40... Interwoven control section, 43A... D/A converter,
45...Soft control latch, 46...
Interconnected control latch.

Claims (2)

[Claims] (1) In a processing device for an electronic musical instrument in which a microcomputer itself generates musical tones under program control, a first latch means for latching a digital musical tone signal generated by the microcomputer at the timing of a program control signal from the microcomputer; a second latch means provided between the output of the first latch means and the input of the digital-to-analog converter means, and latches the output signal from the first latch means at accurate sampling period signal timing; An electronic musical instrument processing device comprising: (2) 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.