JPH02125297A - Digital sound signal generating device - Google Patents

Abstract

PURPOSE:To enable FM and AM without requiring any signal source dedicated to modulation by supplying the output of one of pitch converting means as a control signal to another pitch converting means and obtaining a frequency- modulated digital sound signal. CONSTITUTION:The output of one pitch converting means 20 is supplied as the control signal to another pitch converting means so that digital sound signals are outputted through pitch converting means 20A, 20B, and 20C respectively, thereby obtaining the frequency-modulated digital sound signal from the latter pitch converting means. Consequently, the need to provide the signal source dedicated to the modulation is eliminated, each sound source signal can be FM-modulated and AM-modulated, and a variety of reproduced sounds are obtained.

Description

[Detailed description of the invention]

The present invention will be explained in the following order. A. Field of industrial application B. Overview of the invention C. Prior art D. Problem to be solved by the invention E. Means for solving the problem (Fig. 1) F. Effect G. Example Gl. Overall structure of the example ('fJJ5 Figure) G2 Configuration of the main parts of the embodiment (Figs. 1 and 2) G, Structure of other main parts of the embodiment F & (Fig. 3) G, Operation of the main parts of the embodiment (Figs. 1 and 2)
Figure 2) G, Operation of other essential parts of the embodiment (Figures 3 and 4) H Effect of the invention A: Industrial application field The present invention is a digital audio signal generator suitable for electronic musical instruments, etc. Regarding. B. Summary of the Invention The present invention provides a digital audio signal generation device in which a plurality of digital audio signals are outputted via pitch conversion means or amplitude control means, in which the output of one of the pitch conversion means or amplitude control means is used as a control signal. By supplying to other pitch conversion means or amplitude control means and obtaining a digital audio signal subjected to frequency modulation processing or amplitude modulation processing, there is no need to provide a signal source dedicated to modulation, and the configuration of the apparatus can be simplified. It was made so that it could be done. 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. Note that noise may be used as a sound source for sound effects. 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, (7) the sound of each instrument is digitally recorded, this is written in a memo U (ROM), and the signal of the instrument is read out from this memory. The so-called sampler sound sources started to be prized. 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. Also, for each instrument, only the sound signal of a specific pitch (pitch) is written into the memory, and the signal read out from the memory is subjected to pitch conversion processing to obtain the sound signal of the desired pitch. There is. Furthermore, a signal waveform called a formant at the initial stage of sound production, which is unique to each musical instrument, is written into the memory as is, but a portion that becomes a repeating waveform of the fundamental cycle is written for only one cycle, and is read out repeatedly. These signal processes are naturally digital processes, but for the sake of simplicity, they will each be expressed as analog signal processing functions in this specification. By the way, if an attempt is made to convert the above-mentioned sampler sound source to FM in order to obtain a wider variety of sounds, a problem arises in that a dedicated signal source for modulation is required, which complicates the circuit configuration. A similar problem also occurs when attempting to perform amplitude modulation (AM) processing to obtain performance effects. In view of this, an object of the present invention is to provide a digital audio signal generating device that can be converted into FM or AM without requiring a signal source dedicated to modulation. Means for Solving Problem E75 The first invention provides a digital audio signal generating device that outputs a plurality of digital audio signals through pitch converting means, in which the output of one of the pitch converting stages is This digital audio signal generating device supplies a control signal to another pitch converting means and obtains a frequency modulated digital audio signal from the other pitch converting means. A second aspect of the present invention provides a digital audio signal generator configured to output a plurality of digital audio signals through amplitude control means, in which the output of one of the amplitude control means is supplied as a control signal to the other amplitude control means. However, the present invention is a digital audio signal generating device in which a digital audio signal subjected to amplitude modulation processing is obtained from other amplitude control means. F Effect According to this configuration, each sound source signal can be converted to FM or AM without providing a signal source dedicated to modulation, and a variety of
You can get a variety of playback sounds. G. Embodiment Hereinafter, an embodiment of the digital audio signal generating device according to the present invention will be described with reference to FIGS. 1 to 5. G. Overall Structure of Embodiment FIG. 5 shows the overall structure of an embodiment of the present invention. In Figure 5, (1) is an external sound source ROM.
, which was digitally recorded as described above, for example 1
Various 6-bit data of various instruments are instantaneously compressed, bit rate reduced (BRR encoded) to, for example, 4 bits, and stored in blocks. (10) shows the digital signal processing device (DSP) as a whole, including a signal processing section (11) and a register RAM (12).
) is included. Desired data among 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 section (11). This external RAM (14) has a capacity of, for example, 64 kB, and in addition to sound source data, programs for the CPU (13) are also written therein, and are used in a time-sharing manner. Similarly, the register RAM (
12) is also used by both the signal processing unit (11) and the CPU (13) in a time-sharing manner. In signal processing a'B (11), 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, and then processed as necessary. , as mentioned earlier
The mouth-processed digital audio signal, which is subjected to various processes such as R processing and pitch conversion, is supplied to a speaker (3) via a DA converter (2). Structure of main parts of G2 embodiment The structure of main parts of an embodiment of the present invention is shown in FIGS. 1 and 2. In this embodiment, eight voices IIA, 'IB... nH are synthesized into two channels, left and right, and output, and the digital audio signals of each voice and each channel are time-divisionally processed. Although calculation processing is performed, 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)...(
20H) are voice t4A and voice nB, respectively.
. . . A signal processing unit for voice IIH, which uses the sound source selection data SRC, ~h supplied to the terminal (15) of the external RAM (14) to process the sound source data storage unit (14V
) are respectively supplied with desired sound source data read out from the respective sources. The sound source data supplied to the signal processing R (20^) is supplied to the BRR decoder (21) via the switch S Ig, where it is decompressed as described above and sent to the buffer RA.
It is supplied to the pitch conversion circuit (23) via M (22). The switch S has terminals (31a) and (3
2a) Control data KON (key on) and KOF from register RAM (12) (see Figure 5)
(key off) is supplied to control its opening and closing. The pitch conversion circuit (23) also receives pitch control data P(H) 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)
through the terminal (34a) and switch S2a,
Other voice signals, such as voice nH, are supplied. The switch 32m is connected to the resistor! via the terminal (35a). '2 A M (12) to control data F1. l
ON (FM ON) is supplied 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 ADSR (AD
SR control gI]) are connected to terminals (36a) and (
37a), is supplied to the multiplier <26) via control circuits (27) and (28) and a changeover switch 53m. The connection state of switch Sem 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 transmitted to the second and third multipliers (29+
) and (29r), and control data LVL (left volume) and RVL (right volume) from register RAM (12) are respectively supplied to terminals (38
a) and (39a), the multiplier (291)
and (29r) are supplied. Multiplier (26) (instantaneous value of D output 0 [ITX power, terminal (
41a), is supplied to the register RAM (12), and is also supplied to the terminal (34b) of the signal processing unit (20B).
supplied to Peak value ENV of the output of switch S core a
X is connected to the register RAM (12) via the terminal (42a).
). Furthermore, as shown by the broken line, the output of the terminal (41a) of the signal processing l (20^) can be supplied to the terminal (36b) of the signal processing section (20B). Maps of each control data on register RAM (12) are shown in Tables 1 and 2 below. Table 1 Table 2 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. Each register is 8 bits. In FIG. 2, (50L) and (50R> are the left channel and right channel signal processing sections, respectively, and are the second multiplier (20^) of the signal processing section (20^) in FIG.
The output of 291) is sent to the main adder (51m'l) of the left channel signal processing section (50L>
and switch S. is supplied to the sub adder (51ejl') through the third multiplier (2
The output of 9r) is directly supplied to the main adder (51mr) of the right channel signal processing unit (50R>) via terminal TR, and is also supplied to the sub adder (51er) via switch SSa. Similarly, the voice ++ B, n H signal processing section (
Each output of (20B) to (20H) is a left and right channel signal processing unit (50 and (50R) each adder (51mj, (51eff) and (51mr),
(51er). Both signal processing! Switches corresponding to the same voices of (50L) and (50R) Sea, Ssa; sub
, Ssb""54111Ssh has a terminal (6
1a), (6ib) ... (61h) Control data EON,
(Echo On), EONb...EON are supplied and are opened and closed in conjunction with each other. The output of the main adder (511111>) is supplied to the multiplier (52), and control data MVL (main volume) from the register RAM (12) is supplied to the multiplier (511111) 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 (51el) is sent to the adder (54).
, through the left channel echo control (14E1) of the outer 11FB RAM (14) and the buffer RAM (55) to a digital low-pass filter (56), such as a finite impulse response (FIR) filter. Ru. The echo control gE section (14εl) has terminals (63) and (
64), control data ESA (echo start address) and EDL from register RAM(12)
(Echo Daylay) is supplied. The low-pass filter (56) is supplied with coefficient data C3-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) from the register RAM (12) are supplied. The output of the multiplier (58) is supplied to the adder (53) and the main adder (52). It is synthesized with the output of external RAM (14ε1) and (14E1) in FIG.
Similar to the external RAM (14V) in FIG. 1, r) is a part of the external RAM (14) in FIG. 5 mentioned above, and is used in a time-division manner 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. G. Configuration of Other Main Parts of Embodiment FIG. 3 shows the configuration of an arithmetic unit related to FM in an embodiment of the present invention. In FIG. 3, parts corresponding to those in FIG. 1 and FIG. 5 described above are given the same reference numerals. In FIG. 3, (71) is a multiplier, and the bus (71) is a multiplier.
The outputs of the register RAM M (12) and the register RAM M (22) are supplied via the bus (2).
73), the outputs of the ROMs (74) and (75) are supplied. The output of the ROM (76) is supplied to the adder (81) via the bus (77), and also to the multiplier (71).
) is supplied to the adder (81), and the adder (81)
The output of is supplied to the C register (82). register(
82) is sent to the adder (81) via the bus (77).
and an overflow limiter (83)
and level shifter (84), Y, register (8
5), is commonly supplied to the Y register (86) and the Y2 register (87). The outputs of the registers (85) and (87) are supplied to the multiplier (71) via buses (72) and (73), respectively, and the output of the register (86) is led out. Operation of the main parts of the G4 embodiment Next, the operation of the main parts shown in FIGS. 1 and 2 in one embodiment of the present invention will be explained. The sound source data storage section (14V) contains, for example, a piano,
The sound source data of various musical instruments such as saxophone, cymbal, etc. are stored with numbers 0 to 2550, and the eight sound source data selected by the sound source selection data SRC, ~h are the signals of each voice. Processing section <20A) ~(2
0H), predetermined processing is performed at hours and minutes 211, respectively. In this embodiment, the sampling frequency f is, for example, 4
4.1 ktlz, and one sampling period (1
/fs), a total of 123 cycles of arithmetic processing are performed for 8 voices and 2 channels. (2) The 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, which indicate "key off" and "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 pitch) Do to D corresponds to the key-on and key-off of each voice mA-"H. This allows the messenger (soft nose\us) to set flag 1" only for the voice that he wants to key-on or key-off.
It is no longer necessary to create a program that temporarily writes bits that will not be changed into a buffer register for each individual note, as was done in the past. As mentioned above, in this embodiment, in order to time-divisionally process the 8 voices from 1 to 8 to H, the pitch conversion circuit (23) performs interpolation calculations based on the input data of 4 samples each before and after. That is, oversampling is performed and pitch conversion is performed at the same sampling frequency Pt f s as the input data. The desired pitch is the control data P (H)
and P (shi). Furthermore, if the lower bit of this P (L) is set to 0, then
Uneven thinning of interpolated data can be avoided,
Fine pitch fluctuations do not occur, and high-quality playback sound can be obtained. Control data FMO! from terminal (35a) When the switch 52a is closed, the audio signal data of, for example, voice "H" supplied to the terminal (34, 1) as described above is changed to pitch control data P (H), P (L).
As a result, the audio signal of voice "A" is frequency-modified m (FM). By this, if the modulating signal is an extremely low frequency of several hertz, for example, vibrato is applied to the modulated signal, In the case of an audio frequency modulation signal, the timbre of the reproduced sound of the modulated signal changes, and it is possible to perform FM using a sampler method without the need for a special sound source dedicated to modulation.
You can get the sound source. Note that the control data FMON is written in an 8-bit register similarly to the above-mentioned KON, and each bit D0 to D7 is
correspond to voices I A and II H, respectively. Furthermore, in order to be able to arbitrarily select modulated and modulated voices, a memory is required to temporarily store the modulated signal. In this embodiment, the hardware implementation is simplified by modulating the next stage voice signal with the previous stage voice signal. Furthermore, the voice selected as the modulation signal is provided with a multiplier, 29
n> and (29r), mi-coating is applied using control data LVL and RVL to prevent overflow of audio data. 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. When the MSB of the control data ADSR is "1", the switch S is in the connection state shown in the figure and ADSR control is performed, and when the MSB of the control data ADSR is "0", the switch 53a is in the connection state shown in the figure. The connection state is reversed, 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 the polyline fade-in mode, the original y=Ao-Bo -exp (-kt) =・
The exponential level increase characteristic of the form = (1) is approximated by a broken line with two types of gradients, steep and gentle, which only requires one calculation. In this case, the slope of the section from 0 to 374 levels and the slope of the 374 level
By selecting the slope of the section ~ Lebel as 4 = 1,
A level increase characteristic of a broken line with a good approximation to equation (1) can be obtained. In exponential fadeout mode, y = Ao -exp (-kt) =
" ... has an exponential level drop characteristic of the form (2). Also, in the case of ADRS control, the signal level increases linearly only in the attack section, and decay, sustain, and I
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 directly according to the parameters specified by the upper and lower 4 bits of control data ADSR (2), and the sustain level and decay and release time lengths are It is set according to the parameter value specified by each 2 bits of data 805R(1). In this embodiment, in order to reduce the number of calculations, as mentioned above,
^In the attack section of the DSR mode, the signal level is designed to rise linearly, but by switching the ADSR mode to envelope mode and making the attack section compatible with the polygonal fade-in mode, there are 3 types of decay, sustain, and release. By making exponential fade-out mode correspond to the section, more natural SR control can be performed manually. When the control circuit (27) is in the direct specification mode, the signal of another voice, for example IIH, is sent from the terminal (41h) of the signal processing section (201 (>) to the terminal (208) of the signal processing section (208).
36a), the audio signal of voice "A" is amplitude-modulated by the audio signal of voice wH in the multiplier (26). Various performance effects such as applying tremolo to the modulation signal can be obtained.The signal output and envelope control input of the multiplier (26) are also supplied to the register RAM (12) from the terminals (41a) and (42a), respectively, and the sample By changing the old signal every cycle, for example, when obtaining a plurality of audio signals with greatly different pitches from sound source data of the same instrument, it is possible to obtain a sound (g) with an arbitrary envelope characteristic different from the predetermined SR pattern. The output signal of the multiplier (26) is multiplied by volume control data LVL and RVL in second and third multipliers (291) and (29r), respectively. Both control data are each signed 8-bit data. For example, l s
When one of the control data having the same sign is increased and the other is decreased over a period of time of approximately ec, a so-called panning effect is obtained in which the sound image of the reproduced sound moves between the left and right speakers. Furthermore, if both control data have different signs, the reproduced sound image can be moved beyond the range between both speakers, and by adding an appropriate device, the reproduced sound image can be localized to the rear. is also possible. Signal processing section (50L) and (50R) in Fig. 2
In ・Switch S aa+ S Sal ~ S
4h+ SSh is the terminal (61a) ~ (61h)
The voice to be echoed is selected by the control data EON (EON. to EONh) from . 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 (51e1) is set in the range of, for example, 0 to 255 m5ec by the control data EDL supplied from the terminal (64) to the echo control unit (14EA). Specified equally for left and right 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). Further, the FIR filter (56) is supplied with a signed 8-bit coefficient C6-01 from the terminal (66), so that the filter (56)
The passage characteristics of are set. The echo signal obtained as described above is processed by a multiplier (58
) is multiplied by the control data EVL in the multiplier (5
In step 2), the main audio signal multiplied by the control data MVL is synthesized by an adder (53). Both control data MVL and EVL are unsigned 8 bits and are mutually independent, and the left and right channels are also independent. 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. G. Operation of other main parts of the embodiment Next, the operation of the main parts shown in FIG. 3 in one embodiment of the present invention will be explained. In the case of frequency modulation, the previous voice, for example “H(7)
Sound l (i number (7) m Time value (OtlTX) yo
and P(H). If the pitch value indicated by the P (L) register is Pm, the pitch value after P1 modulation is expressed as the following equation (3). Pm −P (1+yo) ” ”
” (3) Also, the pitch data (
If the pitch data (slot value) of the next sampling period is SL, then the pitch data (slot value SL+y+) of the next sampling period is expressed as follows.
・(4) From this SLm, the RAM (
22) and the ROM (76), and the input data of the pitch conversion circuit (23) and the pitch conversion filter coefficient are respectively output. The actual operation sequence is as follows. In the case of Fi, lON, the coefficient (1/
2) is output, and in the multiplier (71), this coefficient (1/2) is multiplied by the instantaneous value y0 of the voice NH signal from the Y0 register (85).This multiplication result and the ROM (76) are constant CI/2) and addition B(8
1) and the intermediate value expressed by the following equation (5) is C
It is written to the Y2 register (87) via the register (82). )r++xl/2+1/2-(1+yo)/2 ・
==・(5) Next, this intermediate value and register RAM(
The pitch value P from 12) is multiplied by the multiplier (71), and this multiplication result is combined with the constant from the ROM (76).

[0] is added by the adder (81), and the calculated value expressed by the following equation (6) is written to the C register (82). PX(1+yo)/2+O-Pm/2 ””(6)
Also, the value SL on RAM (22) and RO
The coefficient [:1/2] from M (74) is the multiplier (71)
This multiplication result and the calculated value of equation (6) from the register (82) via the bus (77) are multiplied by the adder (8
1) and is supplied to the level shifter (84) via the register (82) and the like. In this shifter (84), a ×2 level shift is performed, and an output as shown in the following equation (7) is sent to the RAM (2) via the Y register (87).
2). (SL XI/2+ Pm/2) X2- S Lm
...-=(7) Modulated 43 as shown in Figure 4
When the instantaneous value of the modulated signal is yo>0, the instantaneous frequency increases as shown in the figure, and yo<
In the case of O, the instantaneous frequency becomes low as shown in C in the figure. ■ Effects of the Invention As detailed above, according to the present invention, the output of one of a plurality of pitch conversion means or amplitude control means is supplied as a control signal to other pitch conversion means or amplitude control means, and frequency modulation processing is performed. Alternatively, since a digital audio signal subjected to amplitude modulation processing is obtained, there is no need to provide a signal source exclusively for modulation, and a digital audio signal generating device can be obtained that can simplify the configuration of the device.

[Brief explanation of drawings]

FIGS. 1 and 2 are block diagrams showing the configuration of essential parts of an embodiment of the digital audio (i-generator) according to the present invention;
The figure is a block diagram showing the configuration of other main parts of an embodiment of the present invention, FIG. 4 is a waveform diagram for explaining the operation of an embodiment of the present invention, and FIG. 5 is an embodiment of the present invention. FIG. 2 is a block diagram showing the overall configuration of the computer. (10) is the digital signal processing 111i2, (12) is the register RAM, (14V) +iMRf'- storage section, (
14Bj2). (14ε") is the echo control section, (20^), (20B)
---・(20H). (50L), (501) are signal processing units, (22)
is RAM, (23) is pitch conversion circuit, (24), (
25), (27), (28) are control circuits, (2
6), (29j2), (29r), (52),
(57), (58), (71) are multipliers, (51f
flj2), (51+nr) is the main adder, (51
e1). (51er) is a sub adder, (74) to (76) are RO
It is M. Agent Ito Sadado Matsukuma 5zumori Figure 4

Claims (1)

[Scope of Claims] 1. In a digital audio signal generating device that outputs a plurality of digital audio signals through pitch converting means, the output of one of the pitch converting means is used as a control signal to the other pitch converting means. 1. A digital audio signal generating device, characterized in that the digital audio signal is supplied as a frequency modulated audio signal from the other pitch converting means. 2. In a digital audio signal generator configured to output a plurality of digital audio signals through amplitude control means, the output of one of the amplitude control means is supplied as a control signal to another amplitude control means, and A digital audio signal generating device characterized in that a digital audio signal subjected to amplitude modulation processing is obtained from an amplitude control means.