JPH02240695A - Musical tone waveform generator - Google Patents

Abstract

PURPOSE:To faithfully produce natural sounds and to allow the free control of musical tone characteristics by generating such carrier signals and modulation signals from which desired musical tone waveforms are outputted, repeatedly generating the arbitrary signal sections of the modulation signals and changing mixing rates with time in synchronization therewith. CONSTITUTION:Carrier signal generating means 11, 12 and modulation signal generating means 2 generate such carrier signals and modulation signals with which the desired musical tone waveforms are outputted from a output means 17 when the mixing rate of the modulation signals is so controlled as to attain, for example, 1 by a mixing control means 4. The desired musical tone waveforms, such as musical tones of a natural musical instrument, can be obtd. if the mixing rate of the modulation signals is previously set at, for example 1, by the mixing control means 4. The carrier signal generating means 11, 12 and the modulation signal generating means 2 repeatedly generate the arbitrary signal sections of the carrier signals and the modulation signals in such a manner that the arbitrary waveform sections of the musical tone waveforms are repeatedly outputted from the waveform outputting means 17. The looped reproduction is executed in this way and the desired musical tone waveforms, such as musical tones of the natural musical instrument, are obtd.

Description

[Detailed Description of the Invention] [Industrial Application Field] The present invention relates to a musical sound waveform generator for an electronic musical instrument,
More specifically, the present invention relates to a musical sound waveform generator that performs frequency modulation to generate musical sound waveforms having various overtone characteristics.

[Prior Art] With the advancement of digital signal processing technology, the first conventional example of an electronic musical instrument using digital processing has not only been able to generate musical sound waveforms with simple characteristics, but also to generate musical sounds of natural musical instruments, one human or Sounds from the natural world, etc. (hereinafter collectively referred to as natural sounds)
A PCM electronic musical instrument has been realized that can directly sample and store music and play it back at any pitch.

On the other hand, as a second conventional example of an electronic musical instrument that can digitally generate musical sound waveforms with various types of complex characteristics,
There are electronic musical instruments based on the FM method described in Japanese Patent Application Laid-open No. 4-33525 or Japanese Patent Application Laid-open No. 126406/1982. This method is basically: e = A-sin (ωct+r(t) sin ω,
t)...(A) The waveform output e obtained by the calculation formula is made into a musical sound waveform, and the carrier wave frequency ωゎ and the modulation wave frequency ω for modulating it are selected in an appropriate ratio, By setting the modulation index I (t) that can change over time, and also setting the amplitude coefficient that can change over time, it is possible to have complex overtone characteristics and the overtone characteristics over time. It is possible to obtain very unique synthesized sounds that can vary.

Further, as a third conventional example of an improved FM system, there is an electronic musical instrument described in Japanese Patent Publication No. 12279/1983.

This method uses a triangular wave operation instead of the sine operation in equation (A), and has the following formula: e = A −T (a+ I'(t) T (θ)l−
- The waveform output e obtained by the arithmetic expression -(B) is used as a musical sound waveform. Here, T(θ) is a triangular wave function generated by the modulated wave phase angle θ.

Then, by advancing the carrier wave phase angle α and the modulating wave phase angle θ at an appropriate advancing speed ratio, and setting the modulation index 1 (t) and amplitude coefficient A as in the first conventional example, Musical sound waveforms can be synthesized.

[Problem to be solved by the invention]

Against the background of the above-mentioned conventional technology, in recent years there has been a demand for electronic musical instruments to have the ability to dynamically produce sounds ranging from very unique musical tones to natural sounds.

However, the first conventional example, the PCM electronic musical instrument,
Although they are very good at making natural sounds themselves, they are not good at processing natural sounds to create unique tones.

In other words, for example, when you want to change the original sound continuously into a sine wave, etc., you use a digital filter or an analog filter to remove the harmonic components of the original sound to obtain a sine wave, but with a digital filter, the circuit size is small. becomes relatively large, and if it is attempted to change its characteristics using a time function such as an envelope, it is necessary to store filter coefficients corresponding to the filter characteristics in addition to the natural sound data. On the other hand, with analog filters,
There are also problems in that the desired characteristics are difficult to obtain, and it is not possible to perform a time-sharing operation for producing a plurality of musical tones in parallel.

Furthermore, contrary to the above, if you want to continuously change the original sound to a musical tone with a more complex harmonic structure, it is not possible to generate new harmonic components using methods such as removing the harmonic structure of the original sound with the above filter. The problem is that it is impossible.

Further, in the PCM electronic musical instrument, loop playback is generally performed in which a specific waveform section of a natural sound is repeatedly played back from memory in order to save storage capacity or to play a sustained sound. However, simply performing loop playback will not work.
A musical tone with the same characteristics is always reproduced in that waveform section, and even if the amplitude envelope is varied, the tone itself does not change, resulting in a monotonous reproduced sound.

As described above, it is difficult to process natural sounds in PCM-based electronic musical instruments, so it is difficult to add timbre changes during loop playback.

On the other hand, for example, the musical tones of an actual musical instrument such as a piano include not only the fundamental wave component based on the ζ pitch frequency but also harmonic components of multiple frequencies that are integral multiples of the fundamental wave component, and even considerably high-order harmonic components exist. Furthermore, overtone components of non-integer multiples may be included. Further, the proportion of each higher-order overtone varies depending on the type of musical instrument, and various overtone characteristics exist depending on the instrument. In this way, musical tones with rich tonal qualities are generated by the presence of harmonic components unique to each instrument. However, electronic musical instruments based on the FM method, which is the second or third conventional method, are very good at manipulating the overtone structure of the musical tones produced, but as an output, they have the above-mentioned characteristics unique to each instrument. When it is desired to obtain a desired musical tone, it is difficult to optimally set its parameters.

That is, since the second conventional example is based on modulation using a sine wave, the frequency components of the musical tone generated by the equation (8) are concentrated in low-order (low-frequency) overtone components, Even if the modulation index 1 (t) is set to a large value and the modulation is applied deeply, high-order (high-frequency) overtone components do not appear well. Therefore, the second conventional example has a problem in that it is not possible to generate musical tones with a rich quality similar to actual musical tones, and the quality of musical tones that can be generated is limited.

On the other hand, in the third conventional example based on the above formula (B), since the modulation is based on a triangular wave that originally contains many overtones, musical tones in which even high-order harmonic components clearly exist as frequency components can be used. However, if you want to obtain a desired musical tone as an output, the carrier wave phase angle α and the modulated wave phase angle θ in (B) 2 can be changed accordingly.
It is difficult to optimally determine the traveling speed ratio, modulation index 1 (t), amplitude coefficient A, etc. In addition to this, the third conventional example is a method of driving a triangular wave with a triangular wave, so for example, in the process of a musical tone gradually attenuating after it begins to sound, the amplitude decreases in order from high-order harmonic components. However, there is a problem in that it is impossible to realize a process in which there is only a single sine wave component corresponding to the pitch frequency.

An object of the present invention is to make it possible to faithfully produce natural sounds with a small circuit scale, to easily and continuously control the overtone components thereof, and to easily synthesize musical sounds such as a single sine wave. To enable loop playback and to freely control musical tone characteristics during loop playback.

[Means for Solving the Problems] The present invention first includes a carrier signal generating means for generating a carrier signal. The means receives as input a carrier wave phase angle signal in which the phase angle repeats an operation in which the phase angle increases sequentially and linearly over time, for example, and converts it according to a certain function and outputs it as a carrier signal. , a ROM, etc., which receives a carrier wave phase angle signal as an address input.

Note that the characteristics of the output carrier signal will be described later.

Next, it has modulation signal generation means for generating a modulation signal.

The means is a means for inputting, for example, the carrier wave phase angle signal, converting it according to a certain function and outputting it as a modulation signal, and is constituted by a ROM or the like having the carrier wave phase angle signal as an address input. Note that the characteristics of the output modulated signal will be described later.

Further, when mixing the modulated signal with the carrier signal generated from the carrier signal generating means, the mixing ratio of the modulated signal to the carrier signal is controlled between O and an arbitrary mixing ratio, and the mixing ratio of the modulated signal to the carrier signal is controlled between O and an arbitrary mixing ratio, It has a mixing control means for outputting a mixed signal in which the modulated signal is mixed at the mixing rate. The means includes, for example, a multiplier that multiplies the modulation signal output from the modulation signal generation means by a modulation index whose value can vary between 0 and 1, and an output signal of the multiplier and the carrier. Adding the carrier signals generated from the signal generating means,
This is an adder that outputs a mixed signal. In addition, the said mixing ratio may change with time so that it may mention later.

Furthermore, it has a waveform output means for inputting the mixed signal output from the mixing control means and outputting a modulated musical sound waveform, the input and output of which have a predetermined functional relationship. The means is, for example, a decoder that converts the mixed signal according to the predetermined functional relationship and outputs it as a musical sound waveform. Alternatively, it is a ROM etc. that uses the mixed signal as an address input.

In addition to the above configuration, the predetermined functional relationship and the carrier signal are the musical tone waveform generated from the waveform output means when the mixing control means controls the mix ratio of the modulation signal to 0. is a sine wave or a cosine wave of a single frequency.

Further, the carrier signal generating means and the modulating signal generating means are configured to generate a desired musical waveform from the waveform outputting means when the mixing ratio of the modulating signal is controlled by the mixing control means to be a predetermined mixing ratio. generating a carrier signal and a modulation signal such that the waveform output means outputs a carrier signal and a modulation signal, and repeating an arbitrary signal section of the carrier signal and the modulation signal so that the arbitrary waveform section of the musical sound waveform is repeatedly output from the waveform output means. Occur.

The mixing control means temporally changes the mixing ratio in synchronization with the repetitively occurring operation.

In addition to the above configurations, the present invention includes an amplitude envelope control means for temporally changing the amplitude envelope characteristic of the musical sound waveform outputted from the waveform output means in synchronization with the repetitive generation operation. The means multiplies, for example, the musical sound waveform outputted from the waveform output means by an amplitude coefficient whose value can vary over time from O to 1, for example, according to a predetermined amplitude envelope function after the start of the repetitive generation operation. It is a multiplier. Note that the means may be a means for normalizing the amplitude of the output of the waveform output means to match the desired tone waveform.

[For production]

The effects of the present invention are as follows.

The musical sound waveform outputted from the waveform output means basically has characteristics obtained by converting the carrier signal outputted from the carrier signal generation means according to a predetermined functional relationship, and furthermore, the mixing control means modulates the carrier signal. By mixing the signals, a characteristic that the musical sound waveform is modulated by the modulation signal is added.

In this case, the relationship between the predetermined functional relationship in the waveform output means and the carrier signal from the carrier signal generating means is
The relationship is set such that when the mixing ratio of the modulation signal is controlled to be 0 by the mixing control means, the musical sound waveform generated from the waveform outputting means becomes a sine wave or a cosine wave of a single frequency.

Thereby, if the mixing ratio of the modulation signal is set to O in advance by the mixing control means, it is possible to generate a musical sound waveform consisting only of a sine wave or a cosine wave of a single frequency.

Furthermore, the carrier signal generating means and the modulating signal generating means output a desired musical waveform from the waveform outputting means when the mixing ratio of the modulating signal is controlled to be, for example, 1 by the mixing control means. A carrier signal and a modulating signal are generated as shown in FIG. Thereby, if the mixing ratio of the modulation signal is set to 1 in advance by the mixing control means, it is possible to obtain a desired musical sound waveform, such as a musical tone of a natural musical instrument.

Further, the carrier signal generating means and the modulating signal generating means repeatedly generate an arbitrary signal section of the carrier signal and the modulating signal so that the arbitrary waveform section of the musical sound waveform is repeatedly output from the waveform output means. . This enables so-called loop playback.

In this case, the mixing control means sets the mixing ratio to, for example, 1 before the start of the loop reproducing operation, and changes the mixing ratio over time to approach O, for example, after the start of the loop reproducing operation. Even during loop playback, the frequency characteristics of the musical sound waveform can be gradually controlled so that the desired musical sound waveform becomes a state containing only a single sine wave component or a single cosine wave component. Alternatively, by continuously changing the mixing ratio to, for example, 1 or more, it is possible to control a desired musical sound waveform so that a unique musical tone having a more complex overtone structure is produced. As a result, it is possible to generate musical waveforms rich in variation using only the carrier signal and modulation signal corresponding to an arbitrary waveform section. For example, it is possible to save the storage capacity of the carrier signal and modulation signal, and it is also possible to You can control the tone so that it doesn't become monotonous.

Along with the above operation, the amplitude envelope characteristic of the musical sound waveform outputted from the waveform output means is controlled by the amplitude envelope control means so as to be temporally attenuated, for example, in synchronization with the loop playback operation. It is possible to realize a process in which the musical sound waveform gradually attenuates after the start of sound production, as in the case of musical tones.

〔Example〕

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

First, before explaining the loop playback operation that is directly related to the present invention, the basic principle of the present invention will be explained.

FIG. 1 is a diagram showing the basic principle of an embodiment of a musical sound waveform generator according to the present invention.

First, from waveform data ROM 1, periodic data F(
t), phase difference data M(t), and a normalization coefficient K (section) having a constant value for each waveform section (described later) are read out in synchronization with each other. First, phase difference data M (L) (ra
d) is multiplied by the modulation index I in the multiplier 4, and then added to the periodic data F(t) (rad) in the adder 3,
Phase angle data F ( for modulating a sine wave)
t)+I −M(t) is obtained. The same data is sin
sinRO, a ROM memory that stores wave data
A sine wave is modulated from M2 and input to the same ROM as an address signal for reading. The modulation output D(t) read from the sin ROM 2 is multiplied by the normalization coefficient K (section) read from the waveform data ROM 1 for each waveform section in the multiplier 5, and then the waveform output 0UT(
t). Here, the maximum absolute value of the amplitude of the sine wave stored in the sine ROM 2 is normalized to be 1.

The operation of the musical waveform generator shown in FIG. 1 based on the above basic principle configuration will be explained below.

First, the original waveform ORG (t) of a musical sound waveform of a natural musical instrument or the like is divided into waveform sections A-D for each periodic section (pitch period) of the fundamental wave, for example, as shown in FIG. 2(a).

Then, within each waveform section, O(rad) or more 2π(r
The periodic data F(t) read from the waveform data ROM 1 in FIG.
) (rad). Now, set the modulation index I in Figure 1 to O
Then, due to the periodic data F(t) itself, sin
When the sine wave stored in ROM 2 is read out by specifying its phase angle linearly and an output of 5 inches (F(t)) is obtained, each waveform section A-D is Each cycle of sine waves corresponding to a phase angle of 0 to 2π (rad) is read out without modulation. In addition, each waveform section A
-D does not need to correspond exactly to the pitch period; in particular, for musical sounds with weak periodicity such as percussion sounds, one waveform section may be defined as, for example, from an appropriate zero-crossing point (when the amplitude is O) to the zero-crossing point. .

Next, the phase difference data M(L) read from the waveform data ROM 1 in FIG.
) using the phase angle data F(L)+M(t) shown in
When n waves are modulated and read out, the waveform output 0LIT(t) is the original waveform 0RG for one waveform section in which the maximum absolute value of the amplitude is normalized to 1, as shown in FIG. 3(a). (
t) is the data to be read out, as shown in FIG. 3(C).
is shown.

Further, the normalization coefficient K (section) for each waveform section read from the waveform data ROM 1 in FIG.
Since the maximum absolute value of the wave amplitude is normalized to 1, this is a coefficient for returning the amplitude to the final original waveform 0RG(t) for each waveform section.

From the above relationship, 0UT(t) = K(interval) -5in(F(t)+I
・M(L)) ・・(1) ORG(t) =K(section) ・5in(F(t)+M(t)) ・・(2
) It can be seen that there is a relationship. As can be seen from these relationships, for each waveform section, when the value of the modulation index I is set to 1, the desired original waveform 0RG (
t), the original waveform 0RG(
period data F(t) and phase difference data M(t) corresponding to
t) and the normalization coefficient K (interval). How to obtain it will be explained by taking as an example the case of waveform section A of the original waveform 0RG(t) in FIG.

First, the method for deriving the periodic data F(t) has already been explained with reference to FIG. 2(b).

Next, in the waveform section A of FIG. 2(a), the original waveform 0RG(t
) as the normalization coefficient K (section A), and by dividing each amplitude value of the original waveform 0RG(t) in the same section by the normalization coefficient K (section). , the amplitude is normalized to within ±1 as shown in FIG. 3(a). In Fig. 3(a), the maximum value of both the positive and negative absolute values is 1, but only one of the maximum values is 1.
, and the other may be 1 or less.

Next, the normalized original waveform 0R obtained in this way
Using G(t), phase difference data M(t) is obtained by the following processes (1) to (2) (see FIG. 4).

■First, waveform section A of the normalized original waveform 0RG(t)
For any time t0 within, that time 1. Find the amplitude A8 of the normalized original waveform ORG (t) corresponding to (
Figure 4(a)).

■ Find the phase angle ρ at the same position as the amplitude A8 found in step ■ on the sine wave whose maximum absolute value of amplitude is 1 (Figure 4 (G)). In this case, the original waveform 0RG ( The position of time t within t) and the position of phase angle ρ8 within the sine wave are determined to have approximately the same relationship. That is, for example, if the time - is within about 1/4 from the beginning of the waveform section, the phase angle is also within about 1/4 from the beginning, from 0 to π/2.
Make decisions nearby.

■PX F(t
Phase difference data M(t,) corresponding to time t is determined as X) (FIG. 4(C)).

■By changing time t8 over the entire range of waveform section A, the above ■~
The process (2) is repeated 7, and phase difference data M(t) corresponding to each time t of the waveform section A is obtained.

The above processing from ■ to ■ is performed on the original waveform 0RG (L) in Figure 2 (a).
is repeated for each waveform section A-D, and the cycle data F(t), phase difference data M(t), and normalization coefficient K (section) found for each waveform section are stored in the waveform data ROM 1 in FIG. do.

By performing the modulation operation with the value of the modulation index I multiplied by the multiplier 4 in FIG. 1 as 1 together with the above data, the waveform output OUT (t) in FIG.
3 original waveform 0RG(t) can be obtained.

Next, in FIG. 1, by changing the value of the modulation index I multiplied by the multiplier 4, various modulated waveform outputs OUT (t) can be obtained.

First, if the modulation index is l-0, an unmodulated sine wave can be obtained within each waveform section as already shown in FIG. 2(C).

Furthermore, by changing the value of the modulation index ■ to 1.0.1.5.2.0, the modulation output D(t) from the sin ROM 2 in FIG. b), (C
), it is possible to obtain waveforms that are sequentially deeply modulated.

As described above, by changing the value of the modulation index {circle around (2)}, various modulated waveforms can be obtained, from a sine wave to a deeply modulated waveform, centering on the original waveform.

In addition, by changing the modulation index l continuously from the start of sound generation to the end of the sound, a waveform output OUT (t) that changes from a deeply modulated state to an stn wave as the musical sound decays, for example, can be obtained. It also becomes possible.

In addition to the above basic principle, as an operation particularly related to the present invention, by repeating the address data Add(1) in FIG. 1 in a predetermined section, a predetermined waveform section is repeatedly output for the waveform output out Performs so-called loop playback. In this case, by temporally changing the value of the modulation index l in the loop playback section, it is possible to perform loop playback that is rich in variation. The specific configuration will be explained below.

FIG. 6A of No. 1 is a concrete configuration diagram of one embodiment of a musical sound waveform generator based on the basic principle of FIG. 1. In this figure, parts given the same numbers as those in the basic configuration of FIG. 1 have the same functions as in FIG. 1. Note that the waveform data ROM 1 in FIG.
In figure A, it is FM ROMI.

It is composed of two ROMs: and K ROM12.

In Fig. 6A, periodic data F(shi) is transmitted from FM ROMI.
Address data ADD#1 for reading the phase difference data M(t) is output from a generation circuit shown in FIG. 6B, which will be described later.

FM R based on the above address data Add (L)
OMI.

The phase difference data M(t) outputted from the multiplier 4 is multiplied by the modulation index ■ latched in the latch 16,
In adder 3, FM ROM1. It is added to the periodic data F(t) output from . The phase angle data F(L)+I −M(t) obtained by this is the sin ROM
It is input to the same ROM as the read address of No.2. Incidentally, the modulation index ■ is outputted from a generation circuit shown in FIG. 6C, which will be described later, and is sequentially latched into the latch 16 in synchronization with the rising edge of the clock CLK#1.

As a result, the modulated output D(t) is output from the sin ROM 2 to the multiplier 5.

On the other hand, the section identification data IB output from the FM ROMI is sent to a D flip-flop (F/F, the same applies hereinafter) 10 that operates in synchronization with the rising edge of the clock obtained by inverting the clock CLK#1 with the inverter 11. At the same time, it is manually input to the first input of the exclusive OR circuit (same for EOR1 and below) 12. Further, the positive logic output Q of the F/F 10 is input to the second input of the EOR 12. With the above circuit configuration, when there is a change in the value of the section identification data 1B sequentially output from the FM ROM 1, the output of the EOR 12 becomes logic [IJ].

Address data ADD#2, which is the address input to K ROMI□, is sent to the adder 13, selector 14, and latch 1.
In the accumulating unit consisting of 5, each address is sequentially accumulated by one address each time the output of the EOR 12 becomes logical "1", using the first address b as an initial value. Note that the start address b is output from a control unit (not shown), and is output from a start pulse 5.
While TRT is at high level, it is selected by the selector 14 and given to the latch 15 as the initial value of the accumulated value. If the start pulse 5TRT is at a low level, the output of the adder 13 is selected to execute the accumulation operation.

As a result, address data ADD# is stored in K ROM12.
2 is given, and a normalization coefficient K (interval) is output from the K ROM 12 to the multiplier 5.

The multiplier 5 multiplies the modulation output D(t) output from the sin ROM 2 by the normalization coefficient K (interval),
This multiplication result is further multiplied by the envelope value generated from the envelope generator 17 in the multiplier 18 .

The result of this multiplication is latched by the latch 19 in synchronization with the rising edge of the clock CLK#1 inverted by the inverter 20, and is output as a waveform output ot+r(t).

Next, FIG. 6B shows a circuit configuration for generating address data ADD#1 of FIG. 6A.

First, basically address data ADD#1 is sent to adder 7
, an accumulator consisting of a selector 8 and a latch 9 uses the start address a as an initial value, and at the rising edge of CLK#1 input to the latch 9 at the address interval of the pitch data d latched in the latch 6. Accumulated sequentially and synchronously. In this case, the start address a is output from a control section (not shown), and is selected by selectors 25 and 8 (via OR circuit 27 in selector 8) while the start pulse 5TRT is at a high level, and is selected as the initial value of the accumulated value. The value is applied to the latch 9. If the start pulse 5TRT is at a low level, the output of the adder 7 is selected to execute the accumulation operation. Further, the pitch data d is outputted from a control section (not shown) and latched into the latch 6 in synchronization with the rise of the start pulse 5TRT.

On the other hand, in synchronization with the rise of the start pulse 5TRT, the latch 23 latches the start address LS of the loop basic section (described later in FIG. In addition, the latch 21 receives the address LE which is +1 to the final address of the loop basic section outputted from the control unit at the same timing.
is latched.

Then, in the subtracter 22, the address LE launched in the latch 21 is calculated from the accumulated value output from the adder 7.
The value of is subtracted, and the result of this subtraction is decoded by the decoder 26.

The output of the decoder 26 becomes high level when the subtraction result becomes 0 or a negative value, that is, when the accumulated value becomes equal to or exceeds the address LE. As a result, the selector 8 selects the output of the selector 25 via the OR circuit 27, and the selector 25 also selects the start pulse 5TR.
If T is at low level, the output of adder 24 is selected.

At this time, in the adder 24, the subtraction result of the subtracter 22, that is, the accumulated value from the adder 7, is added to the start address LS of the basic loop section latched in the latch 23 at the address L.
The amount exceeding E is added and output to the selector 25. Therefore, the latch 9 latches an address that is folded back from the end of the loop basic section to the beginning by the width of the pitch data d.

After this, the addresses of the loop basic section are accumulated, and when the address LE is exceeded, the loop returns to the beginning of the loop basic section again. In this way, the address data A of the loop basic section
DD#1 is repeatedly output.

Next, FIG. 6C shows a circuit configuration for generating the modulation index I shown in FIG. 6A.

First, before starting the performance, the performer gives a control signal SEL in advance from a control section (not shown) to the selector 33, and as a generation source of the modulation index ■ in the loop basic section, which will be described later.
31 or LFO (low frequency oscillator, same below) in IRO
Select 32.

Then, at the timing when the start pulse 5TRT rises to a high level at the start of the performance, the R3 flip-flop (F/F, hereinafter the same) 28 is reset, and its positive logic output Q becomes a low level. As a result, the selector 29
selects the value 1 and outputs it as the modulation index I.

Subsequently, F/F 2 B is set at the timing when the output of the decoder 26 in FIG. 6B becomes a high level, that is, when the loop reproduction operation starts, and its positive logic output Q becomes a high level. After this, the selector 29 selects the output of the selector 33. At the same time, the address counter 30 or LFO 32 is activated.

Now, when the selector 33 is in the mode of selecting the output of the f ROM 31, the address counter 30 has l RO
The start address X where the modulation index data to be read from M31 is stored is set by a control unit, not particularly shown. Then, based on the startup instruction from the positive logic output Q from the F/F 28, the start address
The address is sequentially incremented in synchronization with the rising edge of the clock CLK#1, and is supplied to the lROM31. As a result, the modulation index data corresponding to the above address is read from the I ROM 31, and the modulation index data corresponding to the above address is read out from the I ROM 31.
9 as a modulation index I. Therefore, the modulation index I has a value of 1 before the start of the loop playback operation, and becomes the modulation index data output from the I ROM 31 after the start of the loop playback operation.

On the other hand, when the selector 33 is in a mode in which the output of the LFO 32 is selected, the LFO 32 starts operating based on the activation instruction by the positive logic output Q from the F/F 2 B, and as a result, the LFO 32 starts operating at the rising edge of the clock CLK#l. LFO data is sequentially outputted in synchronization with , and is outputted as a modulation index ■ via the selector 33 and the selector 29. Therefore, the modulation index ■ is the value before the start of the loop playback operation,
After the start of loop playback operation, the L output from LFO32
Output the FO data value.

In the above configurations of FIGS. 6A, 6B, and 6C,
The clocks CLK#1, CLK#2 and start pulse 5TRT are output from a control section, not particularly shown.

Next, FM ROM1. of FIG. 6A. FIG. 7 shows the structure of the data stored in .

In the figure, a set of waveform data from the start of sound generation to the end of sound is stored in order starting from the first address a, and address l contains one cycle data F(t), one phase difference data M(t), and One-bit section identification data IB is stored in sets. In this case, as the address advances, data at each sampling point of each waveform section A, B, C, D, . . . in FIG. 2(a) is stored. Also, as the waveform section changes from A to B to C to D, etc., the value of each address of the section identification data IB changes to "0" in section A, "IJ" in section B, and "0" in section C. ”, “1” in the interval
, . . . and so on, it changes alternately in units of sections.

This section identification data iB is data for identifying the boundary of the section and updating the address specified in the K ROM 12, as will be described later.

Next, FIG. 8 shows the structure of data stored in the K ROM 12 shown in FIG. 6A.

In the same figure, each waveform section A from the start of sound generation to silence,
Corresponding to B, C, D, . . . , normalization coefficients K (sections) are stored one by one in order from the top address b.

The operation of the embodiment having the above configuration will be described below.

First, the normal playback operation before starting the loop playback operation will be explained according to the operation timing chart of FIG.

At the start of sound generation, a start pulse 5TRT is output from a control section (hereinafter simply referred to as the control section), which is not particularly shown.
However, at the timing of tl in Fig. 9, the logic becomes "1" (hereinafter,
Simply call it "1". The same goes for logic "0". ), and immediately after that, the sound generation operation starts from the timing L2 when the clock CLK#1 becomes "1". In this case, the cycle of clock CLK#1 corresponds to the sampling cycle of musical tone generation, and the clock CLK#1 generated from the control section
K#2 is a clock that has the same cycle as clock CLK#1 and is delayed by 174 cycles from the same clock. The following operation is based on the above two clocks CLK#1 and CLK#2.
controlled according to In addition, the start pulse 5TRT maintains "1" for one cycle from when the clock CLK#1 falls to "0" to the next fall to "0" at the start of sound generation, and after that, when the next pulse starts. "0" until the start of pronunciation
maintain.

First, the explanation will be centered on FIGS. 6A and 6B.

The start pulse 5TRT from the control unit is as shown in Fig. 9(a).
At timing L1 when the logic rises to logic "1" as shown in FIG.
latched to.

Next, the timing t when the start pulse 5TRT is “l”
1 to t4, the selectors 25 and 8 select the start address a from the control section, and this start address a is latched by the latch 9 at timing L2 when the clock CLK#I rises to "1", as shown in FIG. The initial value of address data ADD#1 is determined as shown in (d).

As a result, after a short delay time from t2, FMR
The periodic data p(t), phase difference data M(t), and section identification data IB (see Fig. 7) of the first address a of OM1+ are read out as shown in Fig. 9(e), (k), and (1). It will be done.

Thereafter, the address data ADD#1 output from the latch 9 is fed back to the adder 7, and the pitch data d set in the latch 6 is successively accumulated.

Here, the above cumulative value is calculated at each timing t5, t8, t in FIG. 9(d) when the clock CLK#1 rises to "1".
ll, tl4, tl7, etc., are sequentially latched by the latch 9 via the selector 8 and new address data AD
D# 1, period data F(t) and phase difference data M(t
, ) and section identification data IB (see FIG. 7) are read out as shown in FIG. 9(e), (2), and (1). Note that since the start pulse 5TRT falls to "0" at step 4, the selector 8 selects the output of the adder 7 from L4 onward.

In this case, the instruction to start producing a musical tone is given by the performer pressing any key on the keyboard (not shown), for example, and if the pressed key at that time is a high-pitched key, , pitch data d having a large value is latched into the latch 6 from the control section. As a result, FM ROM1. The address width skipped above becomes large, and a waveform output 0UT(L) with a high pitch period is obtained. Conversely, if the lowest note key is pressed, for example, the value 1 is latched as the pitch data d, and this causes the FM ROM1. Above, each data is read out one address at a time, and the waveform output of the lowest pitch period is 0.
LIT (t,) is obtained.

Among the data read out from the FM ROMI as described above, the phase difference data M(t) is input to the multiplier 4. Now, after the start pulse 5TRT rises to "1", the clock CLK#1 rises to "1" at each timing t2, t5, t8, Lll, t (see FIG. 9).
1 in l4, tl7, etc. A modulation index I having a constant value of 1 is sequentially set in the latch 16 from the selector 29 in FIG. 6C. This is because the F/F 28 in FIG. 6C is reset by the rise of the start pulse 5TRT, and the positive logic output Q becomes low level. Therefore, in the multiplier 4, the phase difference data M(t) is multiplied by the modulation index I.

This output is input to adder 3, where FMROM l
It is added to the periodic data F(t) output from I to obtain phase angle data F(t)+I −M(t).

From the above phase angle data F(t)+I −M(t), si
n ROM2 is accessed and the periodic data F(t)
and output of phase difference data M(t) (Fig. 9(g), (1)
) timings L2, t5, L8, tll, t14,
After a short delay, sin RO
From M2 to the modulated output D(t) as shown in FIG. 9(n)
is output.

On the other hand, in FIG. 6A, the start pulse 5TRT is "
The selector 14 selects the start address b from the control unit between the timings tl and 4 of "1", and the start address b is latched by the latch 15 at the timing L3 when the clock CLK#2 rises to "1". As shown in FIG. 9(i), the initial value of address data A[)D#2 is determined.

This causes KRO to be released after a short delay from L3.
The normalization coefficient K (section A) of the first address b of M12 is read out as shown in FIG. 9(j). Thereafter, the address data ADD#2 output from the latch 15 is fed back to the adder 13, and the logical output value of EOR 12 is "0" or "1".
' are accumulated sequentially. In this case, the above cumulative value is calculated at each timing L6, L9, tlo, t12, t15, t1 in FIG. 9(i) when the clock CLK#2 rises to "1".
8, etc., are sequentially latched into the latch 15 via the selector 14, designated as new address data ADD#2, and the normalization coefficient K (interval) of the corresponding address on K ROMIZ becomes as shown in FIG. 9 (j). It is read out as follows.

Here, the start pulse 5TRT is "0" at t4.
'', the selector 14 selects the output of the adder 13 after t4.

In parallel with the above operation, the clock CLK#1 rises to "1" at each timing L2, L5, t8,
After a short delay from Ell, M14, t17, etc., section identification data IB are sequentially output from the FM ROMI as shown in FIG. 9(e). This data is clock CL
At each timing t4, L7, tlo, t13, t16, t19, etc. in FIG. 9(b) when K#1 falls to r□, it is set to F/F 10 and its positive logic output Q is as shown in FIG. It is determined sequentially as shown in (f). Then, this positive logic output Q and the FM ROM1. Section identification data I from
Exclusive OR with B is calculated in EOR12.

Here, the section identification data IB output from ROMI to F is "0" for each waveform section as shown in FIG.
or rl" are stored alternately, so the waveform section is A.
Each time it changes from B to B, B to C, C to D, etc., it changes from "0" to "1", from "1" to "O", and from "0" to "1". Therefore, the output of EORl2 is rlJ only at the time when the waveform section changes, and rQ at other timings.In the example of FIG. 9, the clock CLK from t16 immediately after t14 when the waveform section changes from A to B
The output of the EOR 12 becomes "1" only while #1 is "1". From this, only within the above-mentioned timing, the adder 13 accumulates the value 1 of the address data ADD#2 from the latch 15, and this accumulated value b+1 is obtained when the clock CLK#2 is "
At timing t15 in FIG. 9(5) when the address data rises to "1", it is latched into the latch 15 as new address data ADD#2. Therefore, clock CLK#2 is "1"
After the timing t15 when the address data ADD#2 rises to
The normalization coefficient K (section B) on 2 is read out as shown in FIG. 9(j).

On the other hand, at other timings when the output of the EOR 12 is "0", the adder 13 does not perform the accumulation operation, so the latch 15 has the same address data AD as l timing before.
D#2 is latched. Therefore, at timings L6, t9, t12, t18, etc. in FIG.
2 to 9 as shown in FIG. 1).

In this way, at the timing when the modulation output D(t) based on the period data F(t) and phase difference data M(L) of the waveform section A from the FM ROMI is read out from the sin ROM 2, the K ROMI The normalization coefficient K (section A) corresponding to the waveform section A is read out from □, and similarly, the normalization coefficient K (section B) corresponding to the waveform section length is read out from the K ROM 12 in the waveform section B, and so on. ,
Corresponding to the output of modulated output 0 (1) of each waveform section, the normalization coefficient K (section) of that waveform section is ROMI
Read from □.

As described above, the clock CLK#1 in Figure 9) rises to "1" at each timing t2, t5, t8.1.
After some delay from 11, t14, t17 etc., sin
From ROM 2, the modulation output D(t
) is output, and in parallel with this, the clock CLK#2 of FIG. 9 etc. rises to "L" at each timing L3, t.
After a short delay from 6, L9, t12, t15, t18, etc., the normalization coefficient K (interval) is output from KROMI□ as shown in FIG. 9(j). The modulation output D(L) and the normalization coefficient K (interval) at each of these timings are multiplied by the multiplier 5, and then further multiplied by the envelope data from the envelope generator 17 by the multiplier 18. Each timing L4, t7, tlo, t13 when the clock CLK#1 falls to "0" in FIG.
At t16, t19, etc., it is latched by the latch 19 as shown in FIG. 9 (0), and the waveform output 0UT at each timing.
(t) is determined. Here, envelope generator 1
7 and the multiplier 18 are not shown in FIG. 1, but they may be operated when it is desired to add an envelope other than the original waveform to the waveform output 0UT(t). For example, as described later, amplitude control during loop playback I do.

By the operation shown above, the first
It is possible to realize the same operation as the musical sound waveform generator shown in the figure.

Here, the overall operation timing including the above operation based on FIG. 9 is shown in FIG. 10.

At Ll, start pulse 5TR is applied as shown in the same figure (b).
After T rises, clock CLK#1 in the same figure (a)
Each time the address data ADD# is output from the generation circuit of FIG. 6B to the FM ROMII of FIG. 6A,
1 increases sequentially as shown in Figure 10(e) (9th
(corresponding to figure (d)). As a result, FM ROM1. From Fig. 10(f), (periodic data F (L
) and phase difference data M(t) are sequentially output as the contents of each address in FIG. 7 in synchronization with the rising edge of clock CLK#1 (more precisely, as shown in FIG. (There is a slight delay from the rise of clock CLK#l)
. Here, in FIG. 10 (each waveform section W, , W2,
W, . . . are the waveform section A in FIG. 2(a) or FIG.
, B, C2D, . . . And the 10th
The periodic data F(t) in FIG. 2(f) corresponds to each waveform section W+, Wz, W:l, . . . as shown in FIG. 2(I)).
The operation in which the phase angle linearly increases from O to 2π is repeated each time.

At the rising edge of each clock CLK#1 in each of the above waveform sections W1, W2, W3, . . . , the value of the modulation index I multiplied by the phase difference data M(t) by the multiplier 4 in FIG. Therefore, the phase angle data F(t)+I −M(t) output from the adder 3 becomes F(t)+M(t), which is expressed in FIG. 3(a) or the equation (2) above. Like, the 6th
The waveform output 0LIT (t) outputted from the sin ROM 2 in Figure A from the latch 19 via the multiplier 5.18 is the desired original waveform ORG (t).

In this case, after a certain waveform section, the performer may wish to perform loop playback in which the last waveform section is repeatedly played back.

If loop playback is performed simply, the outline of the waveform output 0UT(t) will be as shown in FIG. 11(a). That is, by performing loop playback using the waveform section Wn as a basic loop section, playback is performed while preserving the waveform of the waveform section Wfi, as shown in (1) to (2) in the figure. If this is reproduced as is, a waveform output 0tJT(j) of a sustained tone such as an organ can be obtained.

Here, there is a case where it is desired to attenuate the waveform output while loop playback is being performed. To do this, the envelope generator 17 in FIG. 6A outputs envelope data as shown in FIG. 11(b), and the multiplier I8 multiplies it. As a result, the waveform output OUT (t) in FIG. 11(a) is converted as shown in FIG. 11(e).

This makes it possible to generate attenuated tones, but in this case, the musical tone characteristics of the loop playback section from ■ to ■ are as follows:
Since the musical tone characteristics of the loop basic section Wn are repeated, the output becomes monotonous as a whole.

Therefore, in this embodiment, in the loop reproduction section, the modulation index T multiplied by the phase difference data M (L) in the multiplier 4 of FIG. 6A is changed to the value before loop reproduction as shown in FIG. 11 ((f)). 1
Control is performed so that the attenuation occurs sequentially from As explained in FIG. 2(C) or FIG. 5, the modulation index I changes from the original waveform 0RG(t) to a sine wave as the value approaches 0 from 1. Therefore, by controlling the value of the modulation index ■ as shown in FIG. 11(d), the waveform output OUT (L ) can be obtained.

Also, when outputting the sustained tone waveform output o'r(t) by loop playback as shown in FIG. By varying the value of index 1 at a low frequency as shown in FIG. 11(e), it is possible to add variation to the timbre.

The specific operation of the embodiment shown in FIGS. 6A, 6B, and 6C for realizing the above operation will be described below.

First, when the performer intends to perform loop playback using the waveform section W7 in FIG.
FM ROM1 in the figure. The starting address LS of the loop basic section Wn related to the above is latched from a control section (not shown) to the latch 23 in synchronization with the rising edge of the start pulse 5TRT. Further, at the same timing, the latch 21 latches an address LE which is the final address of the loop basic section plus one from the control section.

After the above operation, the address data AD
D#1 is the address interval of the pitch data d latched by the latch 6G with the start address a as the initial value in the accumulator consisting of the adder 7, the selector 8 and the latch 9 in FIG. 6B, and the timing is 1. Thereafter, in synchronization with the rising edge of the clock CLK#1 input to the latch 9, the accumulated values are sequentially accumulated as shown in FIG. 10(e).

In this case, the LEO value latched in the latch 21 is subtracted from the accumulated value output from the adder 7 in the subtracter 22 in FIG. 6B. In FIG. 10(e), the result of the subtraction becomes a positive value and the output of the decoder 26 maintains a low level until the address data ADD#1 reaches the address LE. Thereby, the selector 8 selects the output from the adder 7 and performs the normal accumulation operation described above.

Then, at timing t15 in FIG. 1O, the 6B
The output of the latch 9 in the figure approaches or matches the final address (LE-1) of the loop basic section Wn, and this output is sent to the adder 7.
When the accumulated result becomes equal to or exceeds the address LE, the subtraction result of the subtracter 22 becomes O or a negative value, and the output of the decoder 26 is output immediately after timing tt5 as shown in FIG. 10(C). It becomes high level at timing ttS. As a result, the selector 8 selects the output of the selector 25, and the selector 25 also selects the output at the timing t in FIG.
At step 16, since the start pulse 5TRT is at a low level, the output of the adder 24 is selected.

At this time, in the adder 24, the subtraction result of the subtracter 22, that is, the accumulated value from the adder 7, is added to the start address LS of the basic loop section latched in the latch 23 at the address L.
The amount exceeding E is added and output to the selector 25. Therefore, latch 9 has clock CL immediately after t16.
At timing t17 when K#l rises, the address wrapped from the end of the loop basic section to the beginning with the width of pitch data d is latched. As a result, the address data ADD#l returns to the beginning of the loop basic section as shown in FIG. 10(e). Note that the return address differs depending on the value of the pitch data d. Furthermore, due to the turnaround at t17, the output of the adder 7 becomes lower than the address LE, so the output of the subtracter 22 becomes a positive value again and the decoder 2
Immediately after t17, the output of 6 returns to the low 1z level as shown in FIG. 10(C).

After this, the addresses of the basic loop section are accumulated, and when the address exceeds the address LE, the process returns to the beginning of the basic loop section at timing tlB. In this way, the address data ADD#1 of the loop basic section is repeatedly output (t19, t20, etc. in FIG. 10(e)).

On the other hand, in the modulation index I generation circuit shown in FIG. The value 1 is output as . Then, at timing t16 when the output of the decoder 26 in FIG. 6B becomes a high level, F/F 2 B is set, and its positive logic output Q becomes a high novel. After this, the selector 29 selects the output of the selector 33. At the same time, the address counter 30 or LFO 32
is started.

Now, if the selector 33 is in the mode to select the output of the I RQM 31, the address counter 30 has the i RO
The start address X where the modulation index data to be read from M31 is stored is set by a control unit, not particularly shown. Then, based on the startup instruction from the positive logic output Q from the F/F 28, the start address
The address is sequentially incremented in synchronization with the rising edge of the clock CLK#1, and is supplied to the IROM 31. As a result, the modulation index data corresponding to the above address is read from the IRQ 1431 and output as the modulation index ■ via the selector 33 and the selector 29. Therefore,
The modulation index I has a value of 1 before the start of the loop playback operation, and becomes the modulation index data output from the I ROM 31 after the start of the loop playback operation. And this modulation index data is
At each timing when the clock CLK#1 rises, the 6th A
It is latched by the latch 16 in the figure, and as a result, the timing t
After 17, the value of the modulation index ■ has a characteristic of attenuating as shown in FIG. 10 (5). As a result, the characteristics shown in FIG. 11(d) are realized.

As described above, address data ADD#1 generated in FIG. 6B is first transferred to FM ROM1. in FIG. 6A. By accessing , the period data F(t) and the phase difference data M(t) of the loop basic section W, , can be obtained (see Figure 10).
and () are read out like a comb, and the modulation index 1 is sequentially latched in the latch 16 as shown in FIG. 10 (5). Therefore, the phase angle data F(L)+I generated thereby
-M(t) accesses the sin ROM 2, so that the sin
A modulated output D(t) with waveform characteristics that changes to the wave characteristics is generated.

On the other hand, in FIG. 6C, when the selector 33 is in the mode to select the output of LP032, LP012 starts operating based on the activation instruction by the positive logic output Q from F7F 28, and as a result, LFO32 is activated by clock CLK#1.
LFO data is output sequentially in synchronization with the rising edge of
It is output as a modulation index I via the selector 33 and the selector 29.

Therefore, the modulation index ■ has a value of 1 before the start of the loop playback operation, and outputs the value of the LFO data output from the LFO 32 after the start of the loop playback operation. As a result, although not particularly shown in FIG. 1O, the characteristic of the modulation index I shown in FIG. 11(e) is realized. As a result, the sin ROM in FIG. 6A
From 2 onwards, the loop basic section W.

The modulation output D(t
) is obtained.

During the above loop playback operation, the loop basic section Wn
As shown in FIG. 7, the FM RO of FIG.
The section identification data IB output from the MII takes the same value of either "0" or rl.

Therefore, as long as an address within the loop basic section is specified, the output of the EOR 12 remains at the low level as described above. Therefore, the address data ADD#2 output from the latch 15 maintains the same value, which causes K
The normalization coefficient K (section Wn) corresponding to the loop basic section W7 continues to be output from the ROM1z. Therefore, the sixth
A waveform output whose amplitude is correctly normalized corresponding to the loop basic section Wn is obtained from the multiplier 5 in FIG.

Along with the above operations, by outputting the envelope data with the characteristics as shown in FIG. 11 from the envelope generator in FIG. 6A and multiplying it by the multiplier 8, the amplitude It becomes possible to obtain a waveform output 0UT(t) that changes to a sine wave as the waveform attenuates.

In FIGS. 6A and 6B above, the loop playback section is set to one of the waveform sections defined in FIG. It is also possible to adopt a circuit configuration that performs loop playback across the waveform section, loop playback in an arbitrary section unrelated to the waveform section, etc.

Furthermore, although the original waveform was reproduced with the value of the modulation index I set to 1 before the start of the loop reproduction operation, the value of the modulation index I may of course be changed to another value even in this section. Similarly, after starting the loop playback operation, the I R in FIG. 6C
Although the characteristic of the modulation index (2) outputted from the UM 31 is an attenuating characteristic as shown in Q'1) in FIG. 10, it may also be set to have other characteristics.

On the other hand, the waveform data ROM 1 in Fig. 1 was configured with two ROMs, FM ROMI and K ROM12, but if you can accept that the address specification becomes somewhat complicated, you can configure it with one ROM. It's okay.

Also, in FM ROMI, periodic data F(t)
As shown in Fig. 7, the values for each sampling point are stored, but within one waveform section, the periodic data F
(t) has a linear characteristic in which the phase angle changes from O to 2π (rad) in each waveform interval width as shown in Figure 2 (b), so if there is enough calculation time, By calculating and outputting the data F(t), storage capacity can be saved.

In addition, in FIG. 6A, the waveform output OUT for one musical tone is
(t), but it is also possible to output a plurality of musical sound waveforms in parallel by operating each section in a time-division manner.

The principle structure of one embodiment of the musical sound waveform generator shown in Fig. 1 is as follows.
When modulation index 1 = 0, from equation (1) above, 0UT(
t) =K(interval) -5in(F(t))
...(3), and the periodic data F(t) is shown in Fig. 2(b).
), the data increases linearly within each waveform section, and the normalization coefficient K (interval) is constant within each waveform section, so a sine wave is obtained as the waveform output 0UT(t). Ta.

Also, in the case of modulation index 1=1, as in equation (2) above,
0RG(t) =-K(interval) ・5in(F(t) 10M(t)) ・As (2), original waveform ORG(t
)was gotten. By changing the value of the modulation index I between 0 and 1, the waveform can be continuously changed from the sine wave to the original waveform. Further, if the value of the modulation index I is set to 1 or more, it is possible to continuously change the waveform from the original waveform to a further modulated waveform.

In this way, in this embodiment, at least s
It is characterized by being able to output in waves and original waveforms. Therefore, if a sine wave can be output when I=O and an original waveform can be output when I=1, there is no need to stick to equations (1) and (2), etc., and our(t) = K( interval)・f (
g (F(t,))+I −M(t))・・・・(4
) Any embodiment may be used as long as it implements the calculations in the relationship.

For example, in equation (4) above, let the input be α, let f be r(α)
=(2,/π)α... (O≦α≦π/2) f(α)=-1+(2/π)(3π/2-α)... (
π/2≦α≦3π/2) f(α)=-1+(2/π)(α-3π/2)... (
3π/2≦α≦2π) ...(5) Defined as a triangular wave function satisfying /2) g(β)=π−(π/2) sinβ ・・(π/2≦β≦3π/2) g(β)=2π0(π/2) sinβ ・・(3π/2≦β ≦2π) ... (6) If the value of the modulation index ■ is O, that is, there is no modulation, the above equations (5) and (6) can be transformed into the above (
4) By substituting into the formula, f(g(β))=K(interval L f((π/2) s
in β) = K (interval)・(2/π)(π/2) sin
B=K (interval)・sin β・(0≦β≦π/2) f(g(β))=K(interval)・f((π−(π/2)・
sin β)) = K (interval)・(-1+(2/π)(3π/2-π+(
π/2) sin β)) = K (interval)・st port β・(π/2≦β≦3π/2) f(g(β))=K(interval)・f(2π+(π/2)・
sin β) = K (interval)・(-1+(2/π)(2π++(π/2
)sin β−3π/2))=K(interval)・si口β・(3π/2≦β≦2π)・・・・(7) That is, a single sine wave is output when no modulation is performed.

Also, by substituting the above equations (5) and (6) into the above equation (4),
When the modulation index 1=1, the original waveform [111RG(1
, ), the above-mentioned FIG. 4(b)
The sine wave of is replaced with the triangular wave defined by the above equation (5), and the period data p(t) of the figure (C) is replaced with the above (6
) is replaced with the function g (t) defined by the formula, and the g
Phase difference data M(t) may be determined as difference data from (L).

In this case, as means for generating the triangular wave corresponding to the sin ROM 2 of FIG. 1, it is also possible to generate the triangular wave by a decoder circuit or the like in addition to the ROM.

In addition to the above embodiment, various combinations of functions can be defined as the functions r and g that satisfy the above equation (4).

〔Effect of the invention〕

According to the present invention, if the mixing rate of the modulated signal is set to 0 in advance by the mixing control means, a musical sound waveform consisting only of a sine wave or a cosine wave of a single frequency can be generated, and By setting the mixing ratio to a predetermined mixing ratio (for example, 1), it is possible to obtain a desired musical sound waveform such as the musical tone of a natural musical instrument.

In addition, the present invention allows so-called loop playback,
Even during loop playback, the frequency characteristics of the musical sound waveform can be gradually controlled so that the desired musical sound waveform becomes a state containing only a single sine wave component or a single cosine wave component. Alternatively, by continuously changing the mixing ratio to, for example, 1 or more, it is possible to control a desired musical sound waveform so that a unique musical tone having a more complex overtone structure is produced. As a result, it is possible to generate musical waveforms rich in variation using only the carrier signal and modulation signal corresponding to an arbitrary waveform section. For example, it is possible to save the storage capacity of the carrier signal and modulation signal, and it is also possible to generate a musical waveform during loop playback. You can control the tone so that it does not become monotonous.

Along with the above operation, the amplitude envelope characteristic of the musical sound waveform outputted from the waveform output means is controlled by the amplitude envelope control means so as to be temporally attenuated, for example, in synchronization with the loop playback operation. It is possible to realize a process in which the musical sound waveform gradually attenuates after the start of sound production, as in the case of musical tones.

As described above, the present invention can easily generate both a state in which natural musical tones are produced and a state in which only a single sine wave component or a single cosine wave component is produced, and various overtone characteristics can also be generated. Moreover, it is also possible to change the overtone characteristics during loop playback. The configuration to achieve this can be achieved using only a combination of ordinary ROM, decoder, adder, multiplier, etc., so it is possible to realize complex musical sound waveforms with a simple circuit configuration, and as a result, , it becomes possible to provide high-quality electronic musical instruments and the like at low cost.

[Brief explanation of drawings]

FIG. 1 is a basic principle configuration diagram of an embodiment of a musical sound waveform generator according to the present invention, FIG. 2 is a diagram showing the relationship between ORG (t) and F(t), and FIG. 3 is a diagram showing the relationship between ORG (t) and F(t). t) and F (L) and M(t). Figure 4 is a diagram showing how to obtain M(t). Figure 5 is
FIG. 6A is a diagram showing the relationship between D(t,), F(L), and M(t) when changing . FIG. 6A is a specific configuration diagram of one embodiment of the musical sound waveform generator. FIG. 6B 6C is a configuration diagram of a generation circuit for address data ADD#1, FIG. 6C is a configuration diagram of a modulation index ■ generation circuit, FIG. 7 is a data configuration diagram of FM ROM, and FIG. 8 is a configuration diagram of K ROM.
The data configuration diagram of Figure 9 is 1. An operation timing chart of a specific configuration of one embodiment of a musical sound waveform generator. FIG. 10 is an operation timing chart during a loop playback operation, and FIG. 11 is an explanatory diagram of the principle of operation during a loop playback operation. 1,... FM ROM, 2-... sin ROM, 3.7.13.24... Adder, 4.5.18... Multiplier, 6.9.15.16.19.21 .23...Latch, 8.14.25.29.33...Selector, 10...
・D-flip-flop (F/F), 11.20・ ・
- Inverter, 12... Exclusive OR circuit (EOR), 17... Envelope generator, 22... Subtractor, 26... Decoder, 27... OR circuit, 28... R3-flip-flop (F/F)30...
・Address counter, 31...I ROM. 32...Low frequency oscillator (LFO), ADD#1, ADD#2...Address data,
F(t)...period data, M(t)...phase difference data, ■...modulation index, F(t)+I・M(t,)...phase angle data, IB・
... section identification data, K (section) ... normalization coefficient, D (t) ... modulation output, Ot! T(t)...Waveform output. CLK#1, CLK#2...Clock, 5TR
T...start pulse,

Claims (1)

[Claims] 1) A carrier signal generating means for generating a carrier signal, a modulating signal generating means for generating a modulated signal, and a method for mixing the modulated signal with the carrier signal generated from the carrier signal generating means. Mixing control means that controls a mixing ratio of the modulated signal to the carrier signal between 0 and an arbitrary mixing ratio, and outputs a mixed signal in which the carrier signal and the modulated signal are mixed at the mixing ratio; waveform output means for outputting a modulated musical waveform by inputting the mixed signal output from the mixing control means, the input and output of which have a predetermined functional relationship, the predetermined functional relationship and the carrier signal; is such that when the mixing rate of the modulation signal is controlled to be 0 by the mixing control means, the musical sound waveform generated from the waveform outputting means becomes a sine wave or a cosine wave of a single frequency. and the carrier signal generating means and the modulating signal generating means are configured to generate a desired signal from the waveform outputting means when the mixing control means controls the mixing ratio of the modulating signal to a predetermined mixing ratio. generating a carrier signal and a modulating signal such that a musical sound waveform is outputted, and generating an arbitrary signal of the carrier signal and the modulating signal so that an arbitrary waveform section of the musical sound waveform is repeatedly output from the waveform output means. 2. A musical sound waveform generating device, wherein a section is repeatedly generated, and the mixing control means temporally changes the mixing ratio in synchronization with the repetitive generating operation. 2) The musical sound waveform generator according to claim 1, further comprising an amplitude envelope control means for temporally changing the amplitude envelope characteristic of the musical sound wave outputted from the waveform output means in synchronization with the repetitive generation operation. Device.