JPH02129696A - Electronic musical instrument - Google Patents

Abstract

PURPOSE:To realize the reproduction of a multi-recorded sound by a small number of signal processing circuits by putting the signal processing output of a non-interval component and the signal processing output of an interval component together at the time of the reproduction and reproducing the non- interval component and interval component continuously. CONSTITUTION:This musical instrument is provided with sound signal processing means 20A, 20B to 20H and a storage means 14V which is with pieces of musical sound information divided into the non-interval component and interval component, one sound signal processing means 20A performs the signal processing of one non-interval signal, and the remaining sound signal processing means 20B - 20H performs the signal processing of one interval component. When the sound is reproduced, the output signal of the sound signal processing means 20A and the output signals of the sound signal processing means 20B - 20H are put together and the non-interval component and interval component are reproduced continuously. Consequently, the reproduction of the multi- recorded sound is realized by the processing means 20A - 20H.

Description

[Detailed description of the invention] The present invention will be explained in the following order.

A Industrial field of application B Summary of the invention Means for solving the problem to be solved by the conventional technical invention (Fig. 1) Working example Gl Overall structure of the example (Fig. 3) G2 example Structure of essential parts (Figs. 1 and 2) G, Operation of the embodiment G, Description of other embodiments (Fig. 5) H Effect of the invention A Industrial application field The present invention is applicable to, for example, a combiator. The present invention relates to an electronic musical instrument suitable for use in a sound effect generating device for an electronic game machine.

B. Summary of the Invention The present invention provides an electronic musical instrument suitable for use, for example, in generating sound effects for electronic game machines, in which a plurality of pieces of musical tone information are divided into non-pitch components and pitch components, and the non-pitch components are divided into non-pitch components. The tones are processed by about one system of processing means, and the pitch components are processed by different systems of processing means for each musical tone, so that good multiple tones can be reproduced with a small number of systems of processing means.

C. Prior Art Conventionally, as a sound source for electronic musical instruments or sound effects for game machines, for example, a square wave signal is supplied to a plurality of preset frequency dividers each having a different frequency division ratio and duty ratio, and is output from each frequency divider. There was one that synthesized individual sound source signals (so-called voices) at an appropriate level. As the original oscillation waveform, a triangular wave, a sine wave, etc. are also used.

Furthermore, for some instruments, such as pianos and drums, the entire sound generation period is divided into four sections: attack, decay, sustain, and release, 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 in the same way.

On the other hand, as a sound source for electronic musical instruments, so-called FM is a method in which a sine wave signal is frequency modulated (FM) with a low frequency sine wave signal.
Sound sources are known, and by making the degree of modulation a function of time, it is possible to obtain a wide variety of audio signals (herein audio signals) with a small number of sound sources.

In addition, noise (white noise, etc.) is used as the sound source for sound effects.
is sometimes used.

D. Problems to be Solved by the Invention In order to reproduce the sounds of various real musical instruments using the so-called electronic sound source as described above, extremely complex signal processing is required, which poses the problem of increasing the circuit scale. there were.

Recently, in order to solve this problem, so-called samplers have been developed that digitally record the sounds of various real instruments, write this to memory (ROM), wipe it, and then read out the signal of the desired instrument from this memory. The sound source is now awarded as a prize.

In this sampler sound source, in order to save memory capacity, the digital audio signal is compressed and written to the memory, and the compressed digital signal read from the memory is decompressed and restored to the original digital audio signal.

In this case, only the sound signal of a specific pitch (pitch) for each instrument is written into memory, and the signal read out from the memory is subjected to pitch conversion processing to obtain the fundamental frequency signal of the sound of the desired pitch. That's what I do.

Furthermore, the signal waveform called formant at the beginning of sound, which is unique to each musical instrument (for example, in the case of a piano, the sound of the sound from when the third part is struck until the hammer hits the string) is written to memory and read out as is. However, the part that becomes the repetitive waveform of the fundamental period is written for one period,
Read out repeatedly.

That is, during reproduction, as shown in FIG. 6, a short-time formant component a is followed by a fundamental frequency signal component which is a repetition of the waveform P, and the desired musical instrument sound is obtained. At this time, in the case of a sound such as a piano, the level of the waveform P is gradually lowered according to a predetermined rule, so that a natural musical instrument sound is reproduced.

Incidentally, such electronic musical instruments are generally capable of simultaneously reproducing a plurality of musical instrument sound signals read from a memory, thereby making it possible to reproduce multiple tones. In this case, there may be multiple tones due to multiple pitches of a single musical instrument or multiple tones due to the sounds of multiple types of instruments.

In order to reproduce such multiple tones, it is necessary to have as many signal processing circuits for processing the musical instrument sound signals read from the memory as the number of tones to be multiplexed. Eight sets of signal processing circuits are required.

On the other hand, when using a sampler sound source as described above, if the formant component and the pitch-converted fundamental frequency component are processed separately, separate signal processing circuits are required for each, and two sets are required for each sound. signal processing circuits are required, which has the disadvantage of doubling the number of signal processing circuits.

In view of the above, it is an object of the present invention to provide an electronic musical instrument of this type that can reproduce multiple tones and can be realized using a signal processing circuit with less.

E. Means for Solving the Problems The electronic musical instrument of the present invention includes a plurality of audio signal processing means (2OA), (20B), etc., as shown in FIG. 1, for example.
- (20H) and notes 1, 1 in which multiple pieces of musical tone information are divided into non-pitch components and pitch components and stored! Means (14V)
For example, one sound signal processing means (2OA)
performs signal processing of non-pitch components of a plurality of musical tone information, and each of the remaining sound signal processing means (20B) to (20H) performs signal processing of a pitch component of any one of the plurality of musical tone information. , the output signal of the sound signal processing means (2OA) that performs signal processing of non-pitch components, and the sound signal processing means (20B) that performs signal processing of pitch components during playback.
0H), and the non-pitch component and the pitch component are continuously reproduced.

F Function According to the electronic musical instrument of the present invention, signal processing of non-pitch components of a plurality of pieces of musical tone information is performed by, for example, one sound signal processing means (2OA).
Since the processing is carried out by the processing means (2OA) to (2DH), the reproduction of multiple tones can be realized with a small number of processing means. In this case, since the non-pitch component is a very short-time signal, a plurality of pieces of musical tone information can be processed sequentially.

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

Overall configuration of G1 embodiment FIG. 3 shows the overall configuration of an embodiment of the present invention.

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

(10) shows the digital signal processing device (DSP) as an electronic musical instrument as a whole, and the signal processing section (11)
and register RAM (12). ROM (1)
Desired data among the various sound source data is sent to the CPU (
13) and is transferred to the external RAM (14) via the signal processing unit (11). This external RAM
(14) has a capacity of, for example, 64 kB, and in addition to the sound source data, programs for the CPU (13) are also written and used in a time-sharing manner. Similarly, the register RAM (12) in which various control data etc. are stored is also used for signal processingfffl.
(11) and CPU (13) in a time-sharing manner.

The sound source data read from the external RAM (14) is restored to the original sound source data by BRR decoding processing, which is the reverse of the BRR encoding described above, in the signal processing unit (11). Various types of processing such as SR processing and pitch conversion as described above are performed. The processed digital audio signal is supplied to a speaker (3) via a DA converter (2).

Structure of the main parts of the G2 embodiment The structure of the main parts of the embodiment of the present invention is shown in FIGS. 1 and 2.

In this embodiment, the 8 voices 11A, 11B...lIH are synthesized into two channels, left and right, and output, and the digital audio signals of each voice and each channel are calculated on a time-division basis. However, for convenience of explanation, virtual hardware having the same configuration is provided for each voice and each channel in FIGS. 1 and 2.

In Figure 1, (20^>, (20B)...(
20) 1) are voice 11A and voice m, respectively.
B: A signal processing unit for voice IIH,
The sound source selection data SRC, which is supplied to the terminal (15) of the external RAM (14), is stored in the sound source data storage section (14).
Desired sound source data read from V) is respectively supplied.

In this case, in this example, when reproducing the musical instrument sound stored in the sound source ROM (1) as a non-pitch component and a pitch component, the data of the non-pitch component is stored in the signal processing unit (2OA) of the voice 11A. The pitch component data is supplied to the signal processing units (20B) to (20B) of other voices.
It is controlled by control data to be described later.

The sound source data supplied to the signal processing unit (2OA) is
The data is supplied to the BRR decoder (21) via the switch S0, decompressed as described above, and stored in the buffer RAM.
It is supplied to the pitch conversion circuit (23) via (22). The switch SLM has terminals (31a) and (32
a) Control data KON (key on) and KOF (
key-off) is supplied to control its opening and closing. The pitch conversion circuit (23) also receives pitch control data P(I+) from the register RAM (12) via a control circuit (24) for calculation parameters and the like and a terminal (33a).

P (L) is supplied, and the control circuit (24) is also supplied with signals of other voices, such as voice IIH, via the terminal (34a) and the switch 82m.

The switch S2m is supplied with control data FMON (FM ON) from the register RAM (12) via the terminal (35a) to control its connection state.

The output of the pitch conversion circuit (23) is supplied to the multiplier (26), and control data ENV (envelope control) and ^DSR (ADS) are supplied from the register RAM (12).
R control) are connected to terminals (36a) and (37a), respectively.
, control circuits (27) and (28), and a changeover switch S. and is supplied to the multiplier (26). Switch S. The connection state of is controlled by the most significant bit of control data ADSR.

Note that when noise is used as a sound effect source, although not shown in the figure, for example, the output of an M-series noise generator is switched with the output of the pitch conversion circuit (23) and the multiplier (26)
supplied to

The output of the multiplier (26) is the second and third multiplier (2H)
and (29r), and control data LVL <left volume) and RVL (right volume) from register RAM (12>) are respectively supplied to terminal (38a).
and (39a), the multipliers (291) and (29
r) supplied to

Multiplier (26) (10 output (7) 11 o'clock value 0tlTX
, terminal (41a), register RAM (12>
and the terminal (3) of the 13 signal processing section (20B).
4b). The peak value EIX of the output of the switch S 3a passes through the terminal (42g) to the resistor RA.
It is supplied to M(12).

Furthermore, as shown by the broken line, the output of the terminal (41a) of the signal processing section (2OA) can be supplied to the terminal (36b) of the signal processing section (20B).

Maps of each control data on the register RAM (12) are shown in Tables 1 and 2 below.

Table 2 The control data in Table 1 is prepared for each voice.

The control data in Table 2 is prepared in common for the 8 voices. The control data below address OD relates to FIG. 2, which will be explained below. Note that each register has 8 bits.

In FIG. 2, (50L) and (50R) are signal processing units for the left channel and right channel, respectively, and the output of the second multiplier (29+) of the signal processing unit (2OA) in FIG. Through terminal TL, main adder (51mn) of left channel signal processing section (50L)
through the switch 34a,
It is supplied to the sub adder (51el) and the third multiplier (29
The output of r) is directly supplied to the main adder (51111r) of the right channel signal processing section (50R) via terminal TR, and is also supplied to the sub adder (51er) via switch SSa. Ru.

Similarly, the signal processing unit (20B) of voice 5B-”H
Each output of ~(20H) is connected to each adder (5) of the left and right channel signal processing sections (50L) and (50R).
1ml), (51e1) and (51mr), (51
er).

Switch S4a* SSa corresponding to the same voice of both signal processing units (50L) and (50R): 541
1 ssb...S4h.

Ssh has terminals (61a), (61b)...=
(6th) Control data E ON, (echo on), E ON from register RAM (12)
b...EONh is supplied and opened and closed in conjunction with each other.

In this case, when the signal processing section (2OA>) of voice "A" is performing the above-mentioned signal processing of the non-pitch component, the switch 34a and SSa are controlled so as not to be in the closed state, and the reverberant sound ( echo) is not added.

The output of the main adder (51rnA) is supplied to the multiplier (52), and the control data MVL (tonic 1) from the register RAM (12) is supplied to the multiplier (51rnA) via the terminal (62).
52), and the output of the multiplier (52) is supplied to the adder (52).
3).

On the other hand, the output of the sub adder (51e1) is sent to the adder (54).
, through the left channel echo control section (14112) of the external RAM (14) and the buffer RAM (55).
A digital low pass filter (56), such as a finite impulse response (FIR) filter, is fed. The echo control unit (14E1) receives control data ES from the register RAM (12) via terminals (63) and (64).
A (echo start address) and EDL (echo delay) are supplied.

The low-pass filter (56) is supplied with coefficient data C, ~C1 from the register RAM (12) via a terminal (66).

The output of the low-pass filter (56) is fed back to the adder (54) via the multiplier (57) and is also supplied to the multiplier (58). Both multipliers (57) and (58
) through terminals (67) and (68), respectively.
Control data EFB (
echo feedback) and EVL (echo volume).

The output of the multiplier (58) is supplied to the adder (53), combined with the output of the main adder (52), and then sent to the output terminal L out via the oversampling filter (59).
is derived.

In addition, external RAM (14Bf) and (14E
r) l;! , similar to the external RAM (14V) in FIG. 1, are part of the external RAM (14) in FIG. 3 mentioned above, and are used on a time-division basis for each voice and each channel.

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

Operation of G3 embodiment Next, the operation of one embodiment of the present invention will be explained.

The sound source data storage section (14V) contains, for example, a piano,
Sound source data for various instruments such as saxophones, cymbals, etc. are stored with numbers from 0 to 2550, and sound source data with non-pitch components such as pianos are stored with different numbers for non-pitch components and pitch components. It is stored with . Then, the eight sound source data selected by the sound source selection data SRC, ~, are sent to the signal processing unit (2OA) of each voice.
~(20u), predetermined processing is performed on each in a time-sharing manner.

In this embodiment, the sampling frequency f is, for example, 4
4.1 ktlz, and one sampling period (1
/fs), a total of 128 cycles of arithmetic processing are performed for 8 voices and 2 channels. One calculation cycle is, for example, 170 nSec.

In this example, the start of each voice's sound (key on)
Control of the switches Sla to Slh indicating stop (key-off) is different from normal control and is performed using separate flags. That is, the control data KON (key on) and KOF
(key off) is prepared separately. Both control data are 8 bits each and are written to separate registers. Each bit D. -D corresponds to key-on and key-off of each voice "Δ~"H, respectively.

This allows the user (music software producer) to turn on the key,
All you have to do is set the flag 1'' for only the voice you want to key off, and there is no need for the conventional process of creating a program that temporarily writes bits that will not be changed for each individual note into the Neutral register.

In this embodiment, when reproducing sound source data divided into non-pitch components and pitch components, first read the data of the non-pitch components from the RAM (14V), and then read the data of the non-pitch components from the RAM (14V). As shown in FIG. 4A, the switch SIl& of is controlled to perform signal processing of the non-pitch component a using the voice of this IIA.Then, all data of this non-pitch component a is read out from RAM (14V). Then, the data of only one period of the following pitch component is read out repeatedly, and the switch 5lb of the signal processing section (20B) to (20H) of the empty voice of "B-"H is pressed.
-3lh, and performs signal processing of pitch components using either voice s B or n H. For example, if the ttB voice signal processing section (20B) is vacant, as shown in Figure 4B, this signal processing section (20B)
Then, signal processing is performed on the pitch component A following the non-pitch component A. At this time, the pitch component is the pitch conversion circuit (23)
The data is converted into data with a predetermined pitch.

While reproducing an instrument sound that is composed of the non-pitch component a and the pitch component, for example, when reproducing sounds of the same instrument with different pitches to create a double tone, the non-pitch component a and the pitch component are reproduced. The non-pitch component a', which is the same as the pitch component a, is RAM (14V)
Read from IIA voice signal processing unit (20^)
perform signal processing. At this time, since the pitch component is being processed by the mB voice, the pitch component b' following the non-pitch component a' is replaced by another vacant voice, for example, C in FIG.
As shown in the figure, the voice signal processing section (20C) of the IC
perform signal processing. At this time, the pitch conversion circuit (2
In step 3), the pitch of the pitch component is converted to a different pitch.

Each sound is processed by the left and right channel signal processing units (5
0L).

(50R> main adder (51mA'), (51mr
) or sub adders (51e1) and (51er), and are reproduced as double tones.

In this embodiment, in order to time-divisionally process the 8 voices "A -" H, the pitch conversion circuit (23) performs interpolation calculations, that is, oversampling, based on the manual data of 4 samples each before and after. , the same sampling frequency f as the input data.

Performs pitch conversion. The desired pitch is represented by control data P (H) and P (L).

Note that by setting the lower bit of P (L) to 0, non-uniform thinning of the interpolated data can be avoided, fine pitch fluctuations will not occur, and high-quality reproduced sound can be obtained.

Switch S is activated by control data FMOM from terminal (35a). When the terminal (34a) is closed, the terminal (34a
), for example, the audio signal data of voice IIH is substituted into pitch control data P (H), P (L), and the audio signal to voice # is frequency modulated (FM). .

As a result, if the modulating signal is an extremely low frequency of several hertz, vibrato is applied to the modulated signal, and if the modulating signal is an audio frequency, the timbre of the reproduced sound of the modulated signal changes, and a special FM using sampler method without installing a sound source
You can get the sound source.

Note that the control data FMON is 8 as in the above-mentioned KON.
Each bit Do-D corresponds to a voice II A, u H, respectively.

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

Immediately, control data ADsI) MS B "1" (
7) In the case, the switch 33m is in the connected state shown in the figure and A
When DSR control is performed and the MSB of control data 80SR is "0", switch 33m is connected in the opposite direction to that shown, and envelope control such as fusing is performed.

This envelope control can be directly specified, linear or broken line fade-in, or
Five modes can be selected: linear or exponential fade-out, and the current peak value is adopted as the initial value for each mode.

In addition, in the case of ADR3 control, the signal level increases linearly only in the attack section, and the signal level increases linearly only in the attack section, and
In the 3rd section of J IJ-S, it decreases exponentially.

And the fade-in and fade-out time length is
It is set appropriately for each mode according to the parameter value specified by the lower five bits of the control data ENV.

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

In this embodiment, in order to reduce the number of calculations, as described above,
In the attack section of AOSR mode, the signal level increases linearly, but by switching the ADSR mode to envelope mode and making the attack section compatible with the polygonal fade-in mode, the signal level increases linearly in the attack section, and the signal level increases linearly in the attack section. It is possible to manually perform more natural AO3R control by making it compatible with exponential fade-out mode.

In addition, the signal output of the multiplier (26) and the envelope control are connected to terminals (41a) and (42a), respectively.
By supplying it to the register RAM (12) from An audio signal with envelope characteristics is obtained.

In the signal processing units (50L) and (50R) in FIG. 2, switches S4a+Ssa; ~Ssh
, Ssh receives control data EON (EON) from terminals (61a) to (61h).

~EONh) to select the voice to be echoed. The control data EON is written into an 8-bit register as shown in Table 2 above.

The delay time of the echo given to each voice output from the sub adder (51eJ) can be varied from left to right in the range of 0 to 255 m5ec, for example, depending on the control data EDL supplied from the terminal (64) to the echo control unit (14εl). is specified equally by the channels. Further, the amplitude ratio of the preceding and succeeding echoes is set to be in phase for the left and right channels by signed 8-bit control data EFB supplied from the terminal (67) to the multiplier (57).

Note that the control data ESA from the terminal (63) is
Gives the upper 8 bits of the start address of the part used for echo control in AM (14).

11, F I R7 Iruta (56) 1. :,1;! ,
The signed 8-bit coefficient C3-C1 is supplied from the r1 child (66), and the pass characteristics of the filter (56) are set so that a natural echo sound is obtained to the auditory sense.

The echo signal obtained as described above is processed by a multiplier (58
) is multiplied by the control data EVL in the multiplier (5
The main voice (3) multiplied by the control data MVL in 2)
and an adder (53). Surface control data MVL
and EVL are both unsigned 8 bits and are mutually independent, and the left and right channels are also independent from each other.

Thereby, the levels of the main audio signal and the echo signal can be controlled independently, and it is possible to obtain a reproduced sound field with a rich sense of presence that gives an image of the original sound space.

In this way, according to the electronic musical instrument of this example, the non-pitch component as a formant component is processed by the tlA voice signal processing unit (
2OA) to perform signal processing, and the pitch components are
The signal processing section of the empty voice of either B or llH (
By performing signal processing in steps 20B) to (201(), good musical instrument sounds are reproduced using a sampler sound source that includes non-pitch components of up to 7 double tones using 8 voices. Compared to the case where two voices, a non-pitch component and a pitch component, are produced in proportion, more multiple tones can be reproduced with a smaller number of voices.

Note that in the example shown in Figure 4 above, the same non-pitch component is read out multiple times to create a multiple tone using different bi-sochi sounds of the same instrument, but if the non-pitch component of a different instrument is read out, Of course, multiple sounds can be created by combining the sounds of different instruments.

Also, the two sounds that make up the double sound start almost at the same time, and 1
If it is necessary to read the sound source data of the non-pitch component that makes up the next sound before the readout (playback) of the sound source data of the non-pitch component ends, for example, the processing of the first non-pitch component can be stopped. The next non-pitch component may be processed. Alternatively, a priority order can be set in advance for the non-pitch components of each instrument sound, and when generation is instructed at the same time, only the non-pitch component that can be heard best according to this priority order is processed and produced. good.

By doing this, signal processing can be performed efficiently with a small number of voices even when a plurality of sounds are generated simultaneously, such as a chord.

G. Description of other embodiments In the above embodiment, a case was explained in which non-pitch components were included only in the voice to II, but depending on the music playback situation, it is possible that the voice of this HA alone is insufficient. In such a case, the number of voices to which non-pitch components are assigned may be increased as appropriate. For example, by alternately assigning non-pitch components to the two voices HA and nB, both sounds can be reliably reproduced even when the non-pitch components themselves overlap. An example is shown in Figure 5. Voices HA and #B
and the non-pitch component Xl, X2. The signal processing of X3... is performed alternately, and the vacant parts of the 6 voices of voices "C-"H are used to generate pitch components Yl, Y2. Y3... signal processing may be performed.

Furthermore, it goes without saying that the present invention is not limited to the above-described embodiments, and can take various other configurations without departing from the gist of the present invention.

H Effects of the Invention According to the electronic musical instrument of the present invention, multiple tones can be efficiently reproduced by using signal processing means with a small number of systems (number of voices), and the circuit configuration is simplified.

[Brief explanation of drawings]

1 and 2 are block diagrams showing the configuration of essential parts of an embodiment of an electronic musical instrument according to the present invention, FIG. 3 is a block diagram showing the overall structure of an embodiment of the present invention, and FIGS. FIG. 5 is a route map for explaining the example in FIG. 1, and FIG. 6 is a route map for explaining the sampler sound source. (10) is a digital signal processing device, (12) is a register RAM, (14V) it sound source storage unit, (1
4ε! the law of nature. (14Er) is eco-control unit, (2OA), (20B
>...-(20H). (50L), (50R) are signal processing units, (22)
is RAM, (23) is pitch conversion circuit, (24), (
27), (28) are control circuits, (26>, (291
), (29r), (52), (57), (58
), (71) are multipliers, (51ml), (51mr
) is the main adder, and (51e1), (51er) are the sub adders. Agent Paige Ito
Hidetaka Matsukuma Digital number processing device 10 History example Fig. 3

Claims (1)

[Scope of Claims] A storage means for dividing and storing a plurality of musical tone information into non-pitch components and pitch components, and a plurality of sound signal processing means for processing the musical tone information from the storage means, and at least One of the sound signal processing means performs signal processing of non-pitch components of the plurality of musical tone information, and each of the remaining sound signal processing means
Signal processing of any one of the pitch components of the plurality of musical tone information is performed, and during playback, the output signal of the sound signal processing means that performs the signal processing of the non-pitch component, and the sound that performs the signal processing of the pitch component. An electronic musical instrument characterized in that a non-pitch component and a pitch component are continuously reproduced by synthesizing the output signal of a signal processing means.