JP2734024B2 - Electronic musical instrument - Google Patents

Description

DETAILED DESCRIPTION OF THE INVENTION The present invention will be described in the following order.

A Field of Industrial Use B Outline of the Invention C Prior Art D Problems to be Solved by the Invention E Means for Solving the Problems (FIGS. 1 and 2) F Action G Example G ₁ Overall Example configuration (Figure 3) configuration of the main part of the G ₂ example (FIG. 1, FIG. 2) G ₃ fIELD the present invention on the effect a industrial operations H invention examples, for example, using a computer The present invention relates to an electronic musical instrument suitable for use in a sound effect generator of an electronic game machine.

B SUMMARY OF THE INVENTION The present invention relates to an electronic musical instrument suitable for use, for example, for generating sound effects in an electronic game machine, which divides a plurality of pieces of musical sound information into non-pitch components and pitch components, and delays only this pitch component. In which reverberation processing is performed so that a good reverberation sound can be added.

C Prior Art Conventionally, as a sound source of an electronic musical instrument or a sound source of a sound effect of a game machine, for example, a square wave signal is supplied to a plurality of preset frequency dividers having different frequency division ratios and duty ratios, and output from each frequency divider. Individual sound source signals (so-called voices)
Was synthesized at an appropriate level. As the original oscillation waveform, a triangular wave, a sine wave, or the like is also used.

Further, depending on the musical instrument, the entire tone generation period is divided into four sections of attack, decay, sustain and release, such as a piano or a drum, and the amplitude (level) of the signal exhibits a unique change state in each section. To cope with this, so-called ADSR control is performed so that the signal level of each voice changes similarly.

On the other hand, as a sound source for electronic musical instruments, a so-called FM sound source in which a sine wave signal is frequency-modulated with a low-frequency sine wave signal (FM) is known, and the modulation degree is a function of time. An audio signal (here, meaning an audio signal) can be obtained.

Note that noise (such as white noise) may be used as the sound source of the sound effect.

D Problems to be Solved by the Invention In order to reproduce the sounds of various types of musical instruments using the above-described electronic sound source, extremely complicated signal processing is required, and the circuit scale becomes large. there were.

In recent years, in order to solve this problem, the so-called sampler that digitally records the sounds of various actual instruments, writes them in a memory (ROM), and reads out the signals of the required instruments from this memory. The sound source came to be awarded.

In this sampler sound source, in order to save the capacity of the memory, the digital audio signal is data-compressed and written to the memory, and the compressed digital signal read from the memory is expanded to return to the original digital audio signal.

In this case, only a signal of a sound having a specific pitch (pitch) is written in a memory for each musical instrument, and a signal read from the memory is subjected to pitch conversion processing to obtain a fundamental frequency signal of a sound having a desired pitch. Like that.

Furthermore, the initial signal waveform called a formant, which is unique to each musical instrument (for example, in the case of a piano, a sound such as an operating sound from striking a keyboard to hitting a string) is written and read out to memory as it is. However, the repetitive waveform of the fundamental frequency is written for one cycle,
It is read out repeatedly.

That is, at the time of reproduction, as shown in FIG. 5, a short-time fault cloak component a is obtained followed by a fundamental frequency signal component b which is a repetition of the waveform P, and a desired musical instrument sound is obtained.
At this time, in the case of a sound of a piano or the like, a natural instrument sound is reproduced by gradually lowering the level of the waveform P according to a predetermined rule.

By the way, such an electronic musical instrument is reproduced, but reverberation sound or the like is added to the musical instrument sound so that sounds of various timbres can be reproduced. In particular, when used as a sound effect generator of a game machine, it is often desirable to add reverberation sound in order to obtain a realistic presence.

On the other hand, when the reverberation sound is added to the instrument sound obtained from the sampler sound source as described above, a good reverberation sound is added to the fundamental frequency component. Was added, resulting in an unnatural sound. That is, when a reverberation sound is added by a conventional electronic musical instrument, an effector for performing reverberation processing is connected to an output unit of the electronic musical instrument, and the reverberation sound is added to all output sounds. For example, when playing the sound of a piano, for example, a mechanical operation sound such as a sound of a hammer hitting a string is included as a fault cloak component, but if a process of adding a reverberation sound or the like to such an operation sound is performed, Unnatural sound called so-called flutter echo is added,
There was an inconvenience that the sound was hard to hear.

No conventional electronic musical instrument has solved such a disadvantage when adding reverberation sound.

SUMMARY OF THE INVENTION In view of the above, an object of the present invention is to provide an electronic musical instrument using a sampler sound source that can perform a good reverberation sound adding process.

E Means for Solving the Problem The electronic musical instrument of the present invention is a storage means (14V) for dividing and storing a plurality of musical tone information into a non-pitch component and a pitch component, as shown in FIGS. 1 and 2, for example. And this storage means (14V)
Signal processing means (20
A), (20B)... (20H) and the main addition which leads the output signals of the respective audio signal processing means (20A) and (20B)... (20H) directly to an adder (53) as an output unit Container (51ml),
(51mr) and the output signals of the respective audio signal processing means (20A), (20B)... (20H) via the variable delay means (14El) and (14Er). And second supply means for guiding the sound signal processing means (20A), (20B)... (20H) to output non-pitch component output signals via the first supply means. The audio signal processing means (20A) and (20B) supplied to the output unit
The output signal of the pitch component output by H) is supplied to the output section via the first and second supply means, and the adder (53) as the output section outputs the signal of the first supply means and the second signal. Are combined with the signal of the supply means to provide a reverberation effect only to the pitch component.

F Action According to the present invention, only the pitch component can be changed by the variable delay means (14E
Since l) and (14Er) are supplied to perform reverberation sound adding processing, reverberation sound is not added to non-pitch components, and good reverberation adding processing is performed.

G Embodiment Hereinafter, an embodiment of an electronic musical instrument according to the present invention will be described with reference to FIGS. 1 to 4.

G. Overall Configuration of Embodiment ₁ FIG. 3 shows the overall configuration of an embodiment of the present invention.

In FIG. 3, (1) denotes a sound source ROM such as a ROM cartridge provided externally, and various data of various musical instruments, for example, 16 bits, which are digitally recorded as described above, are quasi-instantaneously compressed. For example, the bit rate is reduced to 4 bits (BRR encoding) and stored in blocks. In this case, in this example, instrument sounds such as piano
A non-pitch component called a formant component at the beginning of pronunciation,
It is stored (stored) separately from a pitch component which is a fundamental frequency signal for one cycle of a sound of a specific pitch.

(10) shows a digital signal processing device (DSP) as an electronic musical instrument as a whole, and includes a signal processing unit (11) and a register RAM (12). Desired data of the various sound source data in the ROM (1) is controlled by the CPU (13) and transferred to the external RAM (14) via the signal processing unit (11). The external RAM (14) has a capacity of, for example, 64 kB, and in addition to the sound source data, a program of the CPU (13) is also written therein, and is used in a time-division manner. Similarly, a register RAM (12) in which various control data and the like are stored is also used by both the signal processing unit (11) and the CPU (13) in a time-division manner.

The sound source data read from the external RAM (14) is processed by the signal processing unit (11) in a BRR that is the reverse of the BRR encoding described above.
After being restored to the original sound source data by the decoding process, various processes such as the above-described ADSR process and pitch conversion are performed as necessary. The digital audio signal after processing is
The signal is supplied to the speaker (3) via the DA converter (2).

A main part of an embodiment of the structure present invention of a main part of the G ₂ embodiment is shown in FIGS. 1 and 2.

^# A in this embodiment, ^# B ...... ^# are respectively adapted to output the synthesized 2-channel left and right 8 voice H, each voice and digital audio signals of each channel by time division, respectively Although arithmetic processing is performed, for convenience of explanation,
1 and 2, virtual hardware having the same configuration is provided for each voice and each channel.

In FIG. 1, the (20A), (20B) ...... (20H) is a respective voice ^# A, a signal processing unit for voice ^# B ...... voice ^# H, terminals of an external RAM (14) (15) _Desired sound source data read from the sound source data storage unit (14V) is supplied by the supplied sound source selection data SRC a to _h .

In this case, in re to play the stored musical instrument sound source ROM (1) is divided into a non-interval component and pitch component this example, data for non-pitch component signal processing portion of the voice ^# A (2
0A) and the pitch component data is controlled by control data described later so as to be supplied to the signal processing units (20B) to (20H) of the other voices.

Sound source data supplied to the signal processing section (20A) via a switch S _1a, is supplied to the BRR decoder (21),
The data is expanded as described above and supplied to the pitch conversion circuit (23) via the buffer RAM (22). Switch S
_1a is connected to the register RA via the terminals (31a) and (32a).
Control data KON (key on) from M (12) (see Fig. 3)
And KOF (key-off) are supplied to control the opening and closing. Further, the pitch conversion circuit (23) is supplied with pitch control data P (H) and P (L) from the register RAM (12) via a control circuit (24) for operation parameters and the like and a terminal (33a). At the same time, the control circuit (24) has a terminal (34a)
And via the switch S _2a, for example, other voice signals, such as voice ^# H is supplied. The switch S _2a, via the terminal (35a), the control data RM from the register RAM (12)
ON (FM ON) is supplied to control the connection state.

The output of the pitch conversion circuit (23) is supplied to the multiplier (26), and the control data ENV from the register RAM (12) is supplied.
(Envelope control) and ADSR (ADSR control) are connected to terminals (36a) and (37a), control circuits (27) and (28), respectively.
Through a switching switch S _3a is supplied to the multiplier (26) and. Connection state of the switch S _3a is controlled by the most significant bits of the control data ADSR.

When noise is used as the effect sound source, although not shown, for example, the omission of an M-sequence noise generator is switched to the output of the pitch conversion circuit (23) and supplied to the multiplier (26).

The output of the multiplier (26) is a second and third multiplier (29l)
And (29r) and register RAM
Control data LVL (left volume) and RVL (right volume) from (12) are supplied to multipliers (29l) and (29r) via terminals (38a) and (39a), respectively.

The instantaneous value OUTX of the output of the multiplier (26) is supplied to the register RAM (12) via the terminal (41a) and to the terminal (34b) of the signal processing unit (20B). Switch S _3a
The peak value ENVX of the output of the register R through the terminal (42a)
Supplied to AM (12).

Further, as indicated by a broken line, the output of the terminal (41a) of the signal processing unit (20A) can be supplied to the terminal (36b) of the signal processing unit (20B).

A map of each control data on the register RAM (12) is shown in Tables 1 and 2 below.

The control data in Table 1 is prepared for each voice. Second
The control data in the table is prepared in common for eight voices. The control data below the address 0D relates to FIG. 2 described below. Each register has 8 bits.

In FIG. 2, reference numerals (50L) and (50R) denote signal processing units for the left channel and the right channel, respectively, for performing reverberation sound (echo) addition processing and the like, and the signal processing unit (20A) shown in FIG. the output of the second multiplier (29l) is, via the terminal Tl _a, left channel signal processing unit mainly adder (50L) (51m
l), and is supplied to a sub-adder (51el) via a switch _S4a, and to a third multiplier (29).
output r) is, via the terminal TR _a, is supplied directly to the right channel signal processing unit mainly adder (50R) (51mr), via a switch S _5a, supplied to the sub adder (51er) Is done.

Hereinafter Similarly, the signal processing portion of the voice ^{# B~ #} H (20B) ~
(20H) output is the left and right channel signal processor (5
0L) and (50R) adders (51 ml), (51el) and (5
1mr) and (51er).

Both signal processing unit (50L), switch S _4a corresponding to the same voice _{(50R), S 5a; S} 4b, S 5b ...... S 4h, the S _5h, the pin (61a), (61b) ...... ( 61h) through the register RAM
From (12), control data EON _a (echo on), EON _b ... EON
_h is supplied and they are opened and closed in conjunction with each other.

In this case, when performing the signal processing of the non-pitch components described above in the signal processing portion of the voice ^# A (20A) is switch S _4a and S _5a is controlled not in the closed state, sound is in the non-pitch components Sound (echo) is not added.

The output of the main adder (51 ml) is supplied to the multiplier (52), and the control data MVL (main volume) from the register RAM (12) is supplied to the multiplier (52) via the terminal (62). , The output of the multiplier (52) is supplied to the adder (53).

On the other hand, the output of the sub adder (51el) is passed through the adder (54), the left channel echo controller (14El) of the external RAM (14) and the buffer RAM (55), for example, a finite impulse response (FIR). Digital reduction filters such as filters (5
6) supplied to. The echo controller (14El) has a terminal (6
Control data ESA (echo start address) and EDL (echo delay) from the register RAM (12) are supplied via 3) and (64).

The reduction filter (56), via a pin (66), the coefficient data C ₀ -C ₇ supplied from the register RAM (12).

The output of the reduction filter (56) is fed back to the adder (54) via the multiplier (57) and is supplied to the multiplier (58). Both multipliers (57) and (58) have
Register RAM via terminals (67) and (68) respectively
Control data EFB (echo feedback) from (12)
And EVL (echo volume).

The output of the multiplier (58) is supplied to an adder (53).
It is combined with the output of the main adder (52), and is led out to the output terminal Lout via the oversampling filter (59).

The external RAMs (14El) and (14Er) in FIG. 2 are each a part of the external RAM (14) in FIG. 3 similarly to the external RAM (14V) in FIG. It is used in a time division manner for each voice and each channel.

The buffer RAM (22) in FIG. 1 and the buffer RAM (55) in FIG. 2 are also used in a time-division manner, as described above.

G. Operation of _Third Embodiment Next, the operation of one embodiment of the present invention will be described.

The sound source data storage unit (14V) stores sound source data of various musical instruments such as pianos, saxophones, cymbals and the like with numbers from 0 to 255, and generates sound source data having non-pitch components such as pianos. Are stored with different numbers assigned to non-pitch components and pitch components. Then, the eight sound source data selected by the sound source selection data SRC a to _h are _processed by the signal processing units (20A) to (20H) of each voice.
Predetermined processing is performed in a time-division manner.

In the present embodiment, the sampling frequency f _s, for example 4
Is selected to 4.1KHz, in one sampling period (1 / f _s) 8
For example, a total of 128 cycles of arithmetic processing are performed on the voice and the two channels. One operation cycle is, for example, 170 nSec
Becomes

In the present embodiment, the control of switch S _1a to S _1h shown starting with (key-on) stops and (key off) to pronounce each voice is different from the normal, is performed using a separate flag. That is, control data KON (key-on) and KOF (key-off) are separately prepared. Each control data is 8
Bits, written to separate registers. Each bit D ₀ to D ₇ of each voice ^# A to I ^# H key-on, corresponding respectively to the key-off.

As a result, the user (music software maker) only needs to set the flag "1" only for the voice to be turned on and off, and a program for temporarily writing the unchanged bit to the buffer register for each individual note, for example, as in the past. The troublesome operation of manufacturing the hologram is not required.

And in this embodiment, re To reproduce the sound source data which is divided into a non-interval component and pitch component, first read from the data of the non-pitch components RAM (14 V), the voice signal processing section of the ^# A (20A ) and controls the switch S _1a of FIG. 4 a
As shown in, performs signal processing of the non-interval component a Voice of the ^# A. The data of the non-pitch component a is stored in the RAM (1
4V) is read out, followed by one of the pitch components
Repeatedly read data by the period, and controls the switch S _1b to S _1h of the signal processing portion of the voice that vacant either ^{# B~ # H (20B) ~} (20H), the ^# B to ^# H The signal processing of the pitch component is performed by one of the voices. For example, when the voice signal processing section of the ^# B (20B) is free, 4th
As shown in FIG. B, the signal processing section (20B) performs signal processing on a pitch component b following the non-pitched component a. At this time, the pitch component b is converted into data of a predetermined pitch by the pitch conversion circuit (23).

During reproduction of the musical instrument sound composed of the non-pitch component a and the pitch component b, for example, when reproducing sounds of different pitches of the same musical instrument to make a heavy tone, as shown in FIG. the same non-pitch components a 'and read from the RAM (14 V) and a, performs signal processing in the voice signal processing section of the ^# a (20A). Since pitch component b in this case is in the signal processing in the voice ^# B Non pitch component a 'lasted pitch component b'
The voice was vacant another, for example as shown in FIG. 4 C, ^#
The signal processing unit (20C) of the voice C performs signal processing. At this time, the pitch is converted to a pitch different from the pitch component b by the pitch conversion circuit (23). Then, the respective sounds are added by the main adders (50 ml) and (51mr) or the sub adders (51el) and (51er) of the signal processing units (50L) and (50R) of the left and right channels to form double sounds. Will be played.

Then, to signal processing in a time division 8 voice ^# A to ^# H in the present embodiment, the pitch conversion circuit (23),
Interpolation operation on the basis of before and after the input data for each four cycles, i.e. subjected to oversampling is performed pitch conversion with the same sampling frequency f _s and the input data. The desired pitch is represented by control data P (H) and P (L).

If the lower bits of P (L) are set to 0, uneven skipping of the interpolation data can be avoided, and a fine pitch fluctuation does not occur, and a high-quality reproduced sound can be obtained.

The control data FMOM from the terminal (35a), when the switch S _2a is closed, is supplied to the terminal (34a) as described above. For example voice ^# H of the audio signal data is the pitch control data P (H), so was assigned to the P (L),
Audio signal of the voice ^# A is a frequency modulated (FM).

Thereby, when the modulation signal is, for example, a very low frequency of several hertz, the modulated signal is vibratoed, and when the modulation signal is of an audio frequency, the tone of the reproduced sound of the modulated signal changes,
Even without special modulation sound source, sampler method FM
A sound source is obtained.

The control data FMON is written into an 8-bit register in the same manner as the above-described KON, and each bit D _{0 to} D ₇ is stored in the voice data.
Corresponding respectively to ^# A to ^# H.

In the multiplier (26), the level of the output signal of the pitch conversion circuit (23) is temporally controlled based on the control data ENV and ADSR.

That is, when the control data ADSR for MSB is "1", switch S _3a is ADSR control is carried out in connected state shown in the figure, the switch S _3a is illustrated in the case of the control data ADSR for MSB is "0" Is in the reverse connection state, and envelope control such as fading is performed.

In this envelope control, direct designation, straight or broken line fade-in,
Five modes of linear or exponential fade-out can be selected, and the current peak value is adopted as the initial value of each mode.

In the case of ADSR control, the signal level rises linearly only in an attack section, and drops exponentially in three sections of decay, cysteine, and release.

The time lengths of the fade-in and fade-out are appropriately set for each mode according to the parameter value specified by the lower 5 bits of the control data ENV.

Similarly, the time lengths of the attack and the sustain are set according to the parameter values specified by the upper and lower 4 bits of the control data ADSR (2).
Decay and release time lengths are defined in the control data ADSR
It is set according to the parameter value specified by each two bits of (1).

In the present embodiment, in order to reduce the number of operations, the signal level rises linearly in the attack section of the ADSR mode as described above. However, the ADSR mode is switched to the envelope mode, and the line is broken in the attack section. In addition to supporting the fade-in mode, the exponential fade-out mode corresponds to the three sections of decay, sustain and release, so that more natural ADSR control can be performed manually.

The signal output and the envelope control input of the multiplier (26) are connected to the terminals (41a) and (42a) through the register RAM, respectively.
Supplied to (12) and rewritten at each sample period, for example, when sound source data of the same musical instrument or a plurality of audio signals having greatly different pitches are obtained, an audio signal having an arbitrary envelope characteristic different from a predetermined ADSR pattern Is obtained.

In the signal processing units (50L) and (50R) in FIG.
Switch _{S 4a, S 5a; S 4h} , are closed respectively by the control data EON from S _5h the terminal (61a) ~ (61h) ( EON a ~EON h), the voice should take the echoes are selected. The control data EON is written into an 8-bit register as shown in Table 2 above.

In this case, when performing the signal processing of the non-pitch components voice ^# A is switch S _4a, S _5a is control data EON _a
, The open state is maintained, and no echo (reverberant sound) is added to the non-pitch component. And ^# B ~ ^# H
When performing pitch component signal processing with
Desired switch by the control data EON _b ~EON _h (switch S _4b, S _5b ~S _4h, either S _5h) is closed, the echo is added to the desired pitch components.

The delay time of the echo added to each voice output from the sub adder (51el) is determined by the control data EDL supplied from the terminal (64) to the echo control unit (14El).
For example, the left and right channels are equally specified in the range of 0 to 255 msec. The amplitude ratio of the preceding and succeeding echoes is set to the same phase in the left and right channels by signed 8-bit control data EFB supplied from the terminal (67) to the multiplier (57).

The control data ESA from the terminal (63) is stored in the external RAM
Of (14), the upper 8 bits of the start address of the portion used for echo control are given.

Further, the FIR filter (56), pin (66) coefficients C ₀ -C ₇ of 8-bit signed is supplied from, auditory, as natural echo sound is obtained, passing the filter (56) The characteristics are set.

The echo signal obtained as described above is multiplied by a multiplier (5
8) Multiplied by the control data EVL in the multiplier (52)
Is synthesized by the adder (53) with the main audio signal multiplied by the control data MVL. Both control data MVL and EVL are both unsigned 8 bits and independent of each other,
The left and right channels are also independent.

As a result, the level of the main audio signal and the echo signal can be controlled independently of each other, and it is possible to obtain a reproduced sound field that is rich in a sense of reality, such as an image of the original acoustic space.

As described above, according to the electronic musical instrument of this example, the non-pitch component as the formant component is converted to the signal processing unit ( ^# 2) of the voice of #A.
To perform the signal processing at 0A), the voice signal processing section vacant one of the pitch component ^{# B~ # H (20B) ~} (20H)
Signal processing, up to 7 with 8 voices
A good instrument sound is reproduced by a sampler sound source including a non-pitch component up to the heavy tone. For this reason, compared with a case where two voices of a non-pitch component and a pitch component are provided for each sound, more multiplex sounds can be reproduced with a smaller number of voices.

The electronic musical instrument of this example has a signal processing unit (50L), (50L).
When performing the reverberation adding process in R), in the case of non-pitch component and pitch component and a different voice-treated instrument sound, a non-pitch components treated with Voice ^# A sub adder ( Five
1el) and (51er) so that reverberation is not added to this non-pitch component. For this reason, a good reverberation sound is added only to the pitch component of the instrument sound,
An instrument sound with a good reverberation sound without an unnatural flutter echo due to a non-pitch component is reproduced.

In addition, the present invention is not limited to the above-described embodiment, and it goes without saying that various other configurations can be adopted without departing from the gist of the present invention.

H Effect of the Invention According to the electronic musical instrument of the present invention, there is an advantage that a musical sound to which a good reverberation sound without flutter echo or the like is added is output.

[Brief description of the drawings]

1 and 2 are block diagrams showing the configuration of the main part of one embodiment of an electronic musical instrument according to the present invention, FIG. 3 is a block diagram showing the overall configuration of one embodiment of the present invention, and FIG. FIG. 1 is a schematic diagram for explaining the example of FIG. 1, and FIG. 5 is a schematic diagram for explaining a sampler sound source. (10) Digital signal processor, (12) Register RA
M, (14V) is a sound source data storage unit, (14El), (14Er) are echo control units, (20A), (20B) ... (20H), (50
L) and (50R) are signal processing units, (22) is a RAM, (23) is a pitch conversion circuit, (24), (27), and (28) are control circuits,
(26), (29l), (29r), (52), (57), (58),
(71) is a multiplier, (51ml) and (51mr) are main adders, (51
el) and (51er) are sub adders.

Claims (1)

(57) [Claims] 1. A storage means for dividing a plurality of musical tone information into a non-pitch component and a pitch component for storage, a plurality of audio signal processing means for processing the musical tone information from the storage means, and the respective audio signals A first supply unit that directly guides an output signal of the processing unit to an output unit; and a second supply unit that delays an output signal of each of the audio signal processing units by a variable delay unit and guides the output signal to the output unit. An output signal of a non-pitch component output by each of the audio signal processing means is supplied to the output section via the first supply means, and an output signal of a pitch component output by each of the audio signal processing means Is supplied to the output unit via the first and second supply units, and the output unit combines the signal of the first supply unit and the signal of the second supply unit to generate a pitch component. Only the reverberation effect Electronic musical instrument to be.