JP3166197B2 - Voice modulator and electronic musical instrument incorporating voice modulator - Google Patents

Description

DETAILED DESCRIPTION OF THE INVENTION

[0001]

BACKGROUND OF THE INVENTION 1. Field of the Invention The present invention relates to an audio modulator which modulates an audio signal with a tone signal to generate a tone having a tone similar to the tone signal.

[0002]

2. Description of the Related Art A conventional electronic musical instrument uses a musical tone based on a pitch (pitch) selected by a performance operator such as a keyboard and a tone selected by a tone selection switch or the like. Occurs.

[0003] Therefore, the tone color of a generated tone is determined by tone color data preselected by a switch or the like, and a switch operation must be performed again to change the tone color during performance. Further, the range of tone colors that can be changed is also limited to the range that can be selected by the tone color selection switch.

Therefore, there is a vocoder in which the timbre of a musical tone signal is changed according to a timbre of a human voice which is changed by the utterance of a performer, and the performance is performed at a pitch determined by a key such as a keyboard operated. However, this vocoder could only be used by those who can play musical instruments such as keyboards.

[0005] In addition, a voice converter for converting the tone of human voice, particularly toys, can easily add an echo,
Conventionally, there has been a device for converting the timbre of voice using a filter having a specific pass frequency band, but it is not suitable for playing music in combination with an electronic musical instrument.

SUMMARY OF THE INVENTION An object of the present invention is to allow a user to sing along with a tune generated by an automatic performance or the like, and to freely change the timbre of the singing voice or the like based on the timbre change of the generated instrument sound. It is possible to make
An object of the present invention is to realize a sound modulation device that is also interesting in terms of "play" or unexpectedness of tone.

[0007]

SUMMARY OF THE INVENTION The present invention basically comprises a modulating means for modulating an external sound signal from a sound detecting means based on a sound signal from a sound detecting means and outputting it as an output sound signal. Having.

More specifically, first, a musical sound input means for inputting a musical sound signal such as a melody sound signal or an accompaniment sound signal generated based on the automatic performance music data, and an audio signal based on the uttering operation of the player, etc. It has audio input means for inputting , for example, a microphone and an amplifier.

Next, there is provided the following modulating means.
That is, the first means divides any one of tone signals such as a melody sound signal and an accompaniment sound signal from the tone detection means into respective tone signals band-limited to a plurality of different frequency bands. Means for dividing the audio signal from the audio detection means into audio signals band-limited to a plurality of different frequency bands the same as the first frequency band division means, and a first frequency band dividing means. An envelope extracting means for extracting each envelope signal from each tone signal from the band dividing means, and a characteristic of each audio signal from the second frequency band dividing means for each of the envelope signals from the envelope extracting means corresponding to the same frequency band. It comprises voltage control varying means for varying the voltage, and accumulating means for accumulating each output from the voltage control varying means and outputting it as an output audio signal. Note that the characteristics of each audio signal may be directly varied by the voltage of each tone signal without providing the envelope extracting means. Alternatively, in addition to the above-described configuration, a configuration may be adopted in which the modulating unit includes a multiplying unit that multiplies a voice signal from the voice detecting unit by a musical tone signal from the musical tone detecting unit.

Here, when the modulation means is configured as an analog circuit, the voltage control variable means is, for example, a voltage control type amplifier .

[0011] The present invention further provides a musical tone from a musical tone input means.
Detects that the signal output level has dropped below a certain level
Level detection means, and the sound signal from the sound input means
Input, and usually the input audio signal is the output sound.
The signal is output as a voice signal, and the level
When the audio signal is detected to be less than
Output as the output audio signal while maintaining a constant level
Level holding means, level holding means and the modulation means
Mixer means for mixing and outputting the output audio signals.
Have. This allows the output of a tone signal from the tone input means.
Voice input means when the level falls below a predetermined level
Output audio signal while maintaining the audio signal from
It is configured to output as

The audio modulator having the above-described configuration may be configured as an independent module, or may be configured to be built in an electronic musical instrument.

[0013]

The tone signal detected by the tone detecting means and the sound signal including the fundamental wave and the overtone component generated by the sound detecting means are divided into a plurality of different frequency bands by the first and second frequency band dividing means. Divided.

[0014] Each audio signal divided into each frequency band is modulated based on the envelope of each tone signal divided into the same band. As a result, an output voice such as a timbre of a musical instrument sound to be automatically played and a singing voice of a player having a nuance is obtained. Also, the level of the tone signal output is 0
The level of the output audio signal becomes 0 when
Can be prevented.

[0015]

DETAILED DESCRIPTION OF THE PREFERRED EMBODIMENTS Four embodiments according to the present invention will be described below with reference to the drawings. First Embodiment <Structure> FIG. 1 is a block diagram showing an overall structure of a first embodiment according to the present invention.

In FIG. 1, a CPU 8 of a tone signal generating circuit 2 for automatic performance includes a removable ROM pack 4 storing note data, and a tone ROM 6 storing tone waveform data and envelope data. Then, the tone data of various melody sounds and accompaniment sounds are read out, and the tone data necessary for the automatic performance are sequentially output to the tone generator 12. The function switch 10 is used for setting a tone color, setting various effects, and the like.

A tone signal composed of a melody tone signal and an accompaniment tone signal is output from the sound source 12 to the modulation circuit 14 based on the tone tone data. Of these two tone signals,
The melody signal is input to the modulation circuit 14 via the signal line A.

The modulation circuit 14 has a signal line B
An audio signal from the audio signal generation circuit 16 based on external audio is input. The audio signal generation circuit 16 includes a microphone 18 and an amplifier 20, and detects a voice emitted by a user.

The modulation circuit 14 modulates the sound signal with a melody sound signal and outputs the sound signal to a sound generation circuit 22 through a signal line C.
Output to The sound generation circuit 22 includes a mixer 24 for mixing a signal from the modulation circuit 14 and an accompaniment sound signal output from the sound source 12 via the signal line D, an amplifier 26 for amplifying an output signal of the mixer 24, and a speaker 28. Consists of

FIG. 2 is a block diagram showing details of the modulation circuit 14 of FIG. In FIG.
Are a plurality of band-specific modulation circuits 14-1, 14-2,.
・ ・ Consists of The band-specific modulation circuit 14-1 includes a band-pass filter (hereinafter referred to as BP) 30-1 to which an audio signal is input, and a BPF 32 to which a melody sound signal is input.
-1. The BPFs 30-1 and 32-2 have the same constant and are configured to pass only signals in the same frequency band.

The BPF 32-1 is connected to an envelope extracting circuit 34-1 for extracting only the envelope signal component of the input melody sound signal, and the BPF 30-1 is a VCA (Voltage) whose amplification is controlled by a control signal. C
ontrolled Amplifier) 36-1.

With such a circuit configuration, the output amplitude of the audio signal input to the BPF 30-1 is controlled by a control signal corresponding to the amplitude of the envelope signal extracted from the melody sound signal input to the BPF 32-1. Is done.

Other modulation circuits 14-2, 14-3,...
Has the same configuration as 14-1, but the constants of the BPFs 30 and 32 are different for each band-specific modulation circuit. Therefore, as shown in FIGS. 3 (a) to 3 (d), each of the band-specific modulation circuits 14-1, 14-2,... Divides the melody sound signal and the audio signal divided for each different frequency band. And the above-described modulation operation is performed.

These band-specific modulation circuits 14-1, 14-2,
Are mixed by mixer 38 and output. <Operation> The melody tone signal and the accompaniment tone signal are separately output from the tone signal generation circuit 2 (FIG. 1) in accordance with a predetermined order based on the tone data for automatic performance, and only the melody tone signal is sent to the modulation circuit 14. Is entered. Then, when the user sings into the microphone 18 in accordance with the accompaniment sound, this audio signal is input to the modulation circuit 14.

The inputted melody sound signal and voice signal are divided into signals of each frequency band by BPFs 30 and 32 in each band-specific modulation circuit. The envelope extraction circuit 3 extracts the melody sound signal divided into the respective frequency bands.
At 4, each envelope signal is extracted.

The VCA 36 uses the envelope signal as a control signal, controls the output level of the audio signal output from the BPF 30 according to the amplitude of the control signal, and outputs it.

The audio signals whose output levels are controlled by the VCA 36 are mixed in the mixer 38 and output to the tone generator 22. As a result, the overtone components of the sound signal generated from the sound generation circuit 22 (FIG. 1) are affected by the complicated envelope change of the melody sound signal in the same frequency band, so that a sound having characteristics very similar to the melody sound is output. This creates a mysterious effect as if the automatic performance device were singing in a human voice.

Next, two other embodiments of the modulation circuit 14 will be described. <Another embodiment of the modulation circuit 14> FIG.
FIG. 14 is a block diagram illustrating a configuration of another embodiment.

In the figure, the output of the BPF 32 is not passed through the envelope extraction circuit 34 shown in FIG. 2 after the output waveform on the negative side is inverted and converted into a pulsating flow in a rectifier circuit (not shown). , Are directly input to the VCA 36 as control signals.

Therefore, for each of predetermined frequency bands (f0 to f1, f1 to f2, f1) as shown in FIGS. 3 (a) to 3 (d).
Each of the audio signals divided into 2 to f3,.
At 36, it is directly controlled by the output waveform itself of the melody signal having the same frequency band as each audio signal.

As a result, a sound different from that of the modulation circuit of FIG. 2 is generated. Other than the circuits described above,
Since it is the same as FIG. 2, the description of its configuration and operation will be omitted.

In the embodiment described above, the magnitudes of the audio signals divided for each predetermined frequency band are all controlled using the VCA 36. Instead of the VCA 36, a VCO (Voltage Controlled Oscillator) or It is also possible to replace with VCF (Voltage Controlled Filter). Although the BPFs 30 and 32 have the same constant, they may have different constants. <Another embodiment of the modulation circuit 14> FIG.
The configuration of still another embodiment is shown.

In the figure, a modulation circuit 14 is constituted by a multiplication circuit, and a voice signal and a melody sound signal are multiplied by a multiplier 44 via signal lines B and A. As a result, a partial tone corresponding to the amplitude of the melody sound signal
And an audio signal having a tone color considerably different from the original audio signal is output.

Next, a second embodiment according to the present invention will be described with reference to FIGS. Note that blocks having the same configuration as in FIG. 1 are denoted by the same reference numerals and description thereof is omitted. Second Embodiment FIG. 6 is a block diagram showing the overall configuration of a second embodiment according to the present invention.

The circuit of FIG. 6 differs from that of FIG. 1 in that the tone signal output from the tone signal generating circuit 2 is a melody tone signal of a musical composition and an accompaniment tone signal such as a chord tone or an obligato tone. And a selection circuit 46 for selecting which signal is used to modulate the audio signal.

In FIG. 6, the output of the sound source 12 is supplied to selection switches 12a, 12b and 12c in a selection circuit 46.
Is applied to either the modulation circuit 14 or the sound generation circuit 22. A melody sound signal is output from the selection switch 12a, and a chord sound signal as an accompaniment sound is output from the selection switch 12b.
An obligato sound signal is also output from 2c as an accompaniment sound.

The contact of the selection switch is connected to the modulation circuit 14
Contact with the signal line A connected to the audio signal generation circuit 1 via the signal line B by one of the above signals.
If the audio signal output from 6 is modulated and brought into contact with the signal line D, the signal is directly supplied to the sound generation circuit 22.

Therefore, as shown in FIG. 6, for example, when the selection switches 12a and 12c are connected to the signal line D and the selection switch 12b is connected to the signal line A,
The audio signal is modulated by a chord sound signal as an accompaniment sound.

As described above, the user appropriately switches the selection switch in the selection circuit 46 to select the melody sound, the chord sound as the accompaniment sound, and the obligato sound.
It is possible to select the desired sound and then modulate the audio signal. Third Embodiment FIG. 7 is a block diagram showing the overall configuration of a third embodiment of the present invention. In the modulation circuit 14 used in the automatic performance apparatus shown in FIGS. 1 and 6, the level of the melody sound signal is changed. Becomes zero, the output level of the audio signal becomes zero.

Therefore, the automatic performance device of FIG. 7 uses a circuit for preventing such a phenomenon from occurring. In the figure, the level detection circuit 48 detects the tone signal generation time 2
When it detects that the melody sound signal output via the signal line A has reached zero level, it outputs a control signal of a certain level to the VCA 50. So VCA50
Secures a predetermined amplification degree.

As a result, the audio signal output from the audio signal generation circuit 16 is output from the VCA 50 at a level based on the predetermined amplification degree even when the melody sound signal becomes zero level.

Note that even when the level of the melody sound signal input as a control signal to the VCA 36 used in FIGS. 2 and 4 becomes zero, for example, a predetermined DC voltage corresponding to a constant control signal is applied to the VCA 36. By giving
A constant level audio signal can be output.

In the first to third embodiments described above, all except for the tone signal generation circuit 2 are constituted by analog circuits. Next, except for the tone generation circuit, all are constituted by digital circuits. A fourth embodiment will be described. Fourth Embodiment <Structure> FIG. 8 is a block diagram showing the overall structure of a fourth embodiment in which the present invention is applied to a keyboard instrument.

In the figure, when a player performs a keyboard operation with a keyboard 78 or a switch operation such as a tone color setting or various effect settings with a function switch 79, the performance information is transmitted to a CPU via a bus 86. (Central processing controller) 70.

The CPU 70 has a ROM (Read Only Memory)
The program stored in the RAM 71 is executed.
AccessMemoy) The performance information is processed using 72 as a work memory. The performance information thus processed, for example, note on / off, velocity, tone color setting data, etc., is sent to the tone generation circuit 76 via the bus 86. The circuit 76 generates musical tones in accordance with the performance information. The tone generation method of the tone generation circuit 76 is as follows.
For example, a PCM system, a modulation system, a harmonic addition system, or the like is used.

Next, a digital tone signal (hereinafter simply referred to as a tone signal) x generated by the tone generating circuit 76.
(n) is input to the DSP 73 through the bus 87 dedicated to the tone signal.

On the other hand, when a song is sung to the microphone 80, an analog audio signal obtained through the amplifier 81 is input to the A / D converter 83 through the low-pass filter 82, and a digital audio signal (hereinafter simply referred to as audio (Referred to as a signal) p (n) and input to the DSP 73. The analog audio signal may be input from the line input terminal LINE IN instead of the microphone.

The DSP 73 stores the tone signal x (n) input from the tone generating circuit 76 and the audio signal p (n) input from the A / D converter 83, and performs a digital filter operation described later. Using a filter coefficient ROM 74 storing various coefficients or a work RAM 75 storing data for digital filter operation, an amplitude modulation process described later is performed.

The digital output tone signal Z (n) obtained by the amplitude modulation processing in the DSP 73 is sent to a D / A converter 77 via a dedicated bus 88, where it is converted into an analog output tone signal. Sound is emitted from the speaker 85 via the amplifier 84.

Next, the configuration and functions of the DSP will be described. <Configuration of DSP> FIG. 9 is an overall configuration diagram of the DSP 73.

As shown in FIG.
Reference numeral 1 denotes a bus 86 connected to the CPU 70, a bus 87 connected to the tone generator 76, a bus connected to the A / D converter 83, and a bus 88 connected to the D / A converter 77.
And each bus is connected to a circuit inside the DSP.

The operation ROM 732 stores the DSP 7
3 is a ROM that stores a microprogram that defines the entire operation, and reads a corresponding program instruction based on a designated address from an address counter 733. The CPU 70 of FIG. 8 sets the data in the address counter 733,
An instruction is given to the address counter 733 as to what program is read from the M.32 and the modulation process described below is executed.

The output of the operation ROM 732 is also supplied to a decoder 734, which outputs various control signals to each circuit in the DSP 73 to perform a desired operation. on the other hand,
The internal bus of the DSP 73 has a filter coefficient ROM of FIG.
74 and a work RAM 75 are connected, and filter coefficients, tone signals x (n), audio signals p (n), etc.
73, or input to and output from the work RAM 75.

The register group 737 includes a plurality of registers for temporarily storing data being operated on, and is connected to each input / output terminal of the multiplier 735 or the adder / subtractor 736 via an internal bus. Then, in order to implement a judging process based on the operation result (comparison result or the like) from the adder / subtractor 736, the address counter 7 via the flag register 738
A flag signal indicating the judgment result is sent to 33.

The address of the address counter 733 is changed according to the output of the flag register 738, and the program instruction is read from the operation ROM 732 according to the address. In this way, a judge process is realized. <Function of DSP> Next, the operation function of the DSP 73 is shown in FIG.
0 will be described with reference to a functional block diagram of FIG.

In the figure, the band-specific modulation sections 89-t (t = 1
, 2,..., N) are the band-specific modulation circuits 14-1,
14-2,..., 14-n
This is realized by the time division processing of the above software. It operates at each sampling period, and at the end of each sampling period, the output from each modulation unit is accumulated by the accumulation unit 94 realized by the software processing of the DSP 73, and the digital output tone signal Z (n) Is output to the D / A converter 77 in FIG.

Each band-specific modulation section 89-t includes band-pass filter sections (BPF sections) 90 and 91 and an envelope extraction section 9
2 and a multiplication unit 93. As will be described later, the BPF units 90 and 91 are realized by a combination of a high-pass filter by software processing common to each band and a low-pass filter by software processing for each band. The multiplication unit 93 is realized by a product-sum operation as described later in combination with the accumulation unit 94.

Next, the BPF units 90 and 91 shown in FIG.
The detailed basic configuration of the envelope extraction unit 92 will be described. 8 and the A / D
Each sampling input from the converter 83
The tone signal x (n) and the audio signal p (n) at each timing n are
By the time division processing of the DSP 73, N BPF units 90
And 91 are filtered.

The BPF sections 90 and 9 of the modulation section 89-t for each band.
1 have the same transfer function Ht (Z). In this embodiment, as shown in FIG. 11, the BPF section is realized by a cascade connection of a high-pass filter section common to each band and a low-pass filter section for each band. in this case,
Assuming that the transfer functions of the high-pass filter unit and the low-pass filter unit are H1 (Z) and H2t (Z), respectively, the transfer function Ht (Z) of the BPF units 90 and 91 is Ht (Z) as shown in FIG.
It is represented by the product of 1 (Z) and H2t (Z).

In the case of the BPF 90 shown in FIG.
(n) is filtered by the high-pass filter of the transfer function H1 (Z), then filtered by the low-pass filter of the transfer function H2t (Z), and the band-limited audio signal Yi (n)
(However, it is output as i = t).

In the case of the BPF 91, the tone signal x
(n) is filtered by the high-pass filter of the transfer function H1 (Z), then filtered by the low-pass filter of the transfer function H2t (Z), and the band-limited tone signal Qj (n)
(However, j = t) is output.

Furthermore, the band-limited musical tone signal Qj
(n) is subjected to processing in the envelope extraction unit 92 in FIG. 10, and this part is, as shown in FIG. 11, a transfer function H _Ej
(Z) is realized by a low-pass filter section having a low cutoff frequency. With such a low-pass filter section, an envelope signal Rj (n) is obtained from the band-limited musical tone signal Qj (n).

Next, the high-pass filter portion of the transfer function H1 (Z), the characteristics of the low-pass filter of the transfer function H2T (Z) and H _Ej (Z), described in detail below. <High-Pass Filter Section of Transfer Function H1 (Z)> FIG. 12 is a block diagram showing the hardware of the high-pass filter H1 (Z) of FIG. This is a second-order FIR digital filter whose transfer function is

[0064]

(Equation 1)

## EQU5 ## In FIG. 12, 95, 96
Of Z ^-1 represents a delay element, 100 and 101 an adder. Reference numerals 97, 98, and 99 denote multipliers, × 2, × 1
, × 1/4 are multiplier coefficients to be multiplied by the multiplier. In the DSP 73 shown in FIGS. 8 and 9, when the filter operation equivalent to the high-pass filter shown in FIG. 12 is the BPF 91 (FIG. 10),

[0066]

(Equation 2)

In the case of the BPF 90 (FIG. 10),

[0068]

(Equation 3)

This is realized by the following discrete operation processing. In this case, since the filter coefficient is a multiple of 2, the multiplication of the coefficient and the signal is realized by simple bit shift processing.

The frequency characteristics of this high-pass filter are as follows:

[0071]

(Equation 4)

Ω = 0 (0 Hz), the gain is minimum,
It has the characteristic that the gain becomes maximum when Ω = π (fs / 2 Hz). Here, fs is the tone signal x (n) and the audio signal p
(n) is the common sampling frequency. FIG. 13 shows the characteristics of this high-pass filter. In the figure, meaningful frequencies are from 0 to fs / 2 Hz. <Transfer function H2t (Z)
FIG. 14 is a block diagram showing the hardware of the low-pass filter H2t (Z) shown in FIG. This is a second-order IIR digital filter whose transfer function is

[0073]

(Equation 5)

Is shown by Then, in this equation, θ and CY
Varies according to the value of the suffix t of each band modulation section 89-t in FIG. 10, and r is a parameter indicating the strength of resonance, that is, the magnitude of the peak, as described later.

In FIG. 14, reference numerals 102 and 103 denote delay elements for delaying one cycle of the sampling clock, and reference numerals 104, 105 and 106 denote multipliers, each of which has a coefficient of 2rCos θ and r ² shown in FIG. , CY. 107 and 108 are adders. DSP
73 (FIGS. 8 and 9), a filter operation equivalent to the low-pass filter having the configuration shown in FIG.

[0076]

(Equation 6)

[0077]

(Equation 7)

This is realized by the following discrete operation processing. Here, the pole of the transfer function is z
= Re ** (jθ), z = Re ** (-jθ) and z =
There is a double zero at zero. This arrangement of poles and zeros,

[0079]

(Equation 8)

FIG. 15 shows the relationship between the pole vector and the zero point vector. As can be understood from the figure, if the angle formed by the line segment connecting the point P moving along the unit circle and the zero point 0 with the real axis is Ω, the pole vector Z ₁ and the zero point vector V at Ω = 0 to π the difference in a length of a vector v ₂ of the _1, initially decreases, then increases to. Vector v
The minimum length of _2, when the point P is closest to electrode Z _1, that is, when theta = Omega.

Here, the frequency Ω of this low-pass filter
The magnitude (amplitude characteristic) of the frequency response at is determined by the ratio between the lengths of the zero point vector v ₁ and the vector v ₂ . Since the value of the zero point vector v ₁ is always 1, the magnitude of the frequency response is proportional to the reciprocal of the magnitude of the vector v ₂ , and becomes maximum at Ω close to θ as described above.
Further, the peak of the magnitude of the frequency response is determined by the value of r, and increases as the value of r approaches 1. FIG.
== − π to π.

On the other hand, the phase of the frequency response is a value obtained by subtracting the angle between the real axis and the vector v ₂ from the angle Ω between the real axis and the zero point vector v ₁ . As apparent from the above description, each band-by-band modulation unit 89-t in FIG. 10, using the center frequency f _t and the sampling frequency f _s of bandwidth, theta = 2 [pi] f
It is determined the value of θ determined in _s / f _t, as shown in FIG. 17, a low pass filter having a peak at each band of the center frequency f _t can be realized.

In this case, the magnitude of the peak varies depending on the value of r as described above. By selecting this value of r, the peak is prevented from affecting the adjacent band, and the value of CY is also determined. Can be set so that the level of the output W _t (n) of each band becomes substantially equal.

The values of r and CY can be obtained, for example, as follows. Now, the center frequency f _{t of} each band
Center frequency f _{t + 1} (= f _t + Δ)
Assuming that the ratio of the magnitude of the frequency response to f) is m,

[0085]

(Equation 9)

By solving a quartic equation of r, as a result, r of 0 <r <1 is selected from the obtained r, and each coefficient −2rcos θ, r ² can be obtained. Numerical results of the calculations, for example, f = 440Hz, f _s = 5KHz
And m = 4,

[0087]

(Equation 10)

Is as follows. The other bands can be obtained in the same manner. The transfer function H ₁ as described above
(Z) and a transfer function H
By connecting the low-pass filters having _2t (z) in a cascade as shown in FIG. 11, the overall transfer function expressed as the product of each transfer function can be used as shown in FIG.
As shown in ( ₁₎ , a pseudo bandpass filter having center frequencies f1 to fn for each band of t = _{1 to n} and a frequency difference Δf between adjacent bands is realized. <Transfer function H
The low-pass filter unit of _Ej (z)> Next, FIG. 19 is a block diagram showing a low-pass filter H _ej of FIG. 11 corresponding to the envelope extraction unit 92 of FIG. 10 (z) in hardware image.

This is because the low-pass filter H
A second-order IIR digital filter of the same form as _2t (z), whose transfer function is

[0090]

[Equation 11]

Is as follows. This is the low-pass filter H
A transfer function of _2t (z), where r = 0.9 and θ = 0. In FIG. 19, the absolute value circuit 109
The absolute value of the output Q _j (n) of the BPF unit 91 in FIG. 10 corresponding to the output W _t (n) of the low-pass filter H _2t (z) is output and sent to the next digital filter unit. 110,
111 is a delay element for delaying one cycle of the sampling clock, and 112, 113 and 114 are multipliers, each of which is multiplied by a coefficient shown in FIG. Also, 11
5, 116 are adders.

In the DSP 73 (FIGS. 8 and 9), a filter operation equivalent to the low-pass filter having the configuration shown in FIG.

[0093]

(Equation 12)

This is realized by discrete operation processing. As shown in FIG. 20, the frequency characteristic of this low-pass filter has a resonance having a maximum value at θ = 0. The cutoff frequency is set to a frequency near the DC component, which is much lower than the cutoff frequency of the low-pass filter H _2t (z) in the lowest band for the purpose of extracting the envelope.

Here, the coefficient CE is for adjusting the output level of each band. FIG.
Converts the envelope signal R _j (n) obtained by the low-pass filter in FIG. 19 into the absolute value | Q of the input signal.
FIG. 5 is a diagram schematically shown in comparison with _j (n) |. FIG.
9, the negative peak value (broken line in FIG. 21) of the input signal is converted into a positive peak value by a calculation corresponding to the absolute value circuit 109, and low-pass filtering is performed. The operation of obtaining the envelope of the musical tone signal Q _j (n) is executed by the envelope extracting unit 92 in FIG.

As described above, the filter function shown in FIGS. 10 to 21 is executed as software processing by the DSP 73 shown in FIG. 8 or FIG. Next, the operation will be described. <Operation of Fourth Embodiment> FIG. 22 is an operation flowchart relating to DSP vocoder processing executed in accordance with the microprogram stored in the operation ROM 732 in the DSP 73 of FIG. 9 for band-pass filter processing and envelope extraction. For low-pass filtering.

By the processing according to the operation flowchart, the band-by-band modulation section 8 of FIG. 10 for each sampling cycle common to the tone signal x (n) and the audio signal p (n).
In 9-t (t = 1 to n), processing corresponding to the BPF units 90 and 91, the envelope extracting unit 92, and the multiplying unit 93, and processing corresponding to the accumulating unit 94 are executed by time division processing. As a result, an output audio signal z (n) is obtained for each sampling period and output to the D / A converter 77 in FIG.

First, the work RAM 75 in FIG. 8 and the register group 737 in the DSP in FIG. 9 are reset as initial settings (step S1). Then, wait for completion of the A / D conversion performed in each period corresponding to the sampling frequency f _s at the A / D converter 83 in FIG. 8 (step
S2) The A / D-converted audio signal p (n) is fetched from the interface 731 (FIG. 9) and sequentially stored in the variable p (n) of the same name in the work RAM 75. At the same time, the tone signal x (n) input from the tone generating circuit 76 (FIG. 8) is fetched from the interface 731 and the work RA
It is sequentially stored in a variable x (n) of the same name in M75 (step S3). Note that each of the three variables p (n), p (n−1), p (n−2), x (n),
It is assumed that x (n−1) and x (n−2) store continuous audio signals and tone signals of the current sample and the past two samples, respectively.

Next, the processing in steps S4 to S7 is performed as shown in FIG.
, And corresponds to the processing of the tone signal x (n) in each of the BPF sections 91 and the envelope extraction section 92 of the band modulation sections 89-1 to 89-N.

First, each variable x from the work RAM 75
(n), x (n−1) and x (n−2) from the register group 737 in FIG.
The tone signals of the current sample and the past two samples are read out to the registers inside the register, and the high-pass filter processing represented by the transfer function H ₁ (z) in FIG.
S4). This processing is a part of the processing corresponding to the BPF unit 91 in FIG.
This is performed using the multiplier 735 and the adder / subtractor 736 in FIG. At this time, each filter coefficient used in the calculation of Expression 2 is read from the filter coefficient ROM 74 (FIGS. 8 and 9). The resulting output is stored in work memory 7
5 is stored in a variable S (n).

The above-described high-pass filter processing is common to each band, and is therefore performed only once. Next,
H _2t which is the rest of the processing corresponding to the BPF unit 91 in FIG.
A low-pass filter process represented by (z) = H _2j (z) and a low-pass filter process represented by a transfer function H _Et (z) = H _Ej (z) in FIG. Is executed in succession.

These processes are performed by the band-specific modulation section 8 shown in FIG.
The processing is repeated as time division processing for N bands corresponding to 9-1 to 89-N. For this purpose, a register j for repetition control for performing time division processing of N bands is provided in the register group 737 of FIG. 9, and is initialized to a value of 1 in step S5. Then, each time the low-pass filter processing for one band is completed in steps S6 and S7, it is determined in step S8 whether or not the content of the register j has reached N. If not, the register j is determined in step S9. Is incremented, and the processing after step S6 is repeated.

This processing is executed by the adder / subtractor 736 and the flag register 738 shown in FIG. 9, and the address counter 733 repeatedly reads out the program instructions corresponding to steps S6 and S7 from the operation ROM 732.

First, the transfer function H shown in FIG. 11 is applied to the contents of the variable S (n) which is the output of the above-described high-pass filter processing.
A low-pass filter process represented by _2t (z) = _H2j (z) is executed (step S6). This processing is based on the above equation (6).
This is an arithmetic process represented by Wt (n) = Qj (n) in the equation (see FIG. 11), and the multiplier 735 and the adder / subtractor 736 in FIG.
Is performed using Each filter coefficient used in the arithmetic processing of Expression 6 is read from the filter coefficient ROM 74 and used for the arithmetic.

The work RAM 75 stores the past 2
Variable Q to store own filter output for sample
j (n-1) and Qj (n-2) are provided.
Reference numeral 37 fetches these contents as needed and uses them in the above calculation. The resulting output is stored in variable Qj (n) in work RAM 75. In addition, each variable Qj (n), Qj (n-1),
And Qj (n-2) are provided for N bands with the subscript j changed.

Next, a low-pass filter process represented by the transfer function H _2t (z) = H _2j (z) in FIG. 11 is performed on the contents of the variable Qj (n) which is the output of the above-described low-pass filter process. Is performed (step S7). This processing is an arithmetic processing represented by the above-described Expression 12, and is performed by the multiplier 735 in FIG.
And using the adder / subtractor 736. Also in this case, each filter coefficient used in the arithmetic processing of Expression 6 is
Read from the filter coefficient ROM 74.

The work RAM 75 stores the past 2
Variable R to store its own filter output for samples
j (n-1) and Rj (n-2) are provided.
Reference numeral 37 fetches these contents as needed and uses them in the above calculation. The output obtained as a result is stored in a variable Rj (n) in the work RAM 75. Note that each variable Rj (n), Rj (n-1),
And Rj (n-2) are provided for N bands with the subscript j changed.

As described above, by the processing of steps S5 to S9, FIG.
BPF unit 9 of band-specific modulation units 89-1 to 89-N of N bands of 0
1, and processing corresponding to the envelope extraction unit 92 is performed.

Subsequently, the processing in steps S10 to S13 is as follows.
Each of the BPF units 90 of the modulation units 89-1 to 89-N in FIG.
For the audio signal p (n). First, each variable p (n), p (n−1),
From p (n−2) to the registers in the register group 737 in FIG.
The audio signals of the current sample and the past two samples are read out, and a high-pass filter process represented by a transfer function H1 (z) in FIG. 11, which is a part of the process corresponding to the BPF unit 90 in FIG. 10, is executed. (Step S10). This processing is an arithmetic processing represented by the above-described equation (2).
35 and an adder / subtractor 736. At this time, each filter coefficient used in the calculation of Expression 2 is read from the filter coefficient ROM 74. The output obtained as a result is stored in a variable S (n) in the work RAM 75.

The above-described high-pass filter processing is common to each band, and is therefore performed only once. Next,
FIG. 1 shows the rest of the processing corresponding to the BPF unit 90 in FIG.
A low-pass filter process represented by the transfer function H _2t (z) = H _2j (z) is performed. This process is repeatedly performed as time-division processing for N bands, corresponding to the band-specific modulation units 89-1 to 89-N in FIG.

For this purpose, a register i for repetition control for performing N-band time-division processing is provided in the register group 737 of FIG. 9, and is initialized to a value of 1 in step S11. Every time the low-pass filter processing is completed, the content of the register i is set to N in step S13.
Is determined, and if not, the step
In S14, the content of the register i is incremented,
The processing after step S12 is repeated.

Also in this case, each circuit of FIG. 9 operates similarly to the case of the register j described above. The processing in step S12 is
This is almost the same as the processing in step S6 described above. That is,
The content of the variable S (n) which is the output of the high-pass filter is subjected to an arithmetic process represented by Wt (n) = Yi (n) in equation (6) (see FIG. 11). At this time, variables Yi (n-1) and Yi (n-2) for storing own filter outputs for the past two samples are provided in the work RAM 75, and the register group 737 stores these variables. Is taken in at any time and used for the above calculation. The resulting output is stored in the variable Yj (n) in the work RAM 75. The variables Yi (n), Yi (n-1), and Yi (n-2) are provided for N bands by changing the subscript i.

As described above, the processing in steps S10 to S13
BPF section 90 of N-band-specific modulation sections 89-1 to 89-N in FIG.
Is performed. The contents of the variables Rj (n) and Yi (n) corresponding to the outputs of the envelope extraction unit 92 and the BPF units 90 and 91 in FIG. 10 are determined by the processing of steps S4 to S14 described above.

Using these contents, like the multipliers 93 of the band-specific modulators 89-1 to 89-N for N bands in FIG.
The following processing corresponding to the processing of the accumulation unit 94 in FIG. 10 is executed.

That is, in step S15, while the contents of the register i = j are changed from 1 to N, Rj
Each multiplication of (n) × Yi (n) is performed by the multiplier 735 of FIG.
It is performed in. The results of these multiplications are shown in FIG.
Are added up using the adder / subtractor 736 of.

The accumulation result obtained in this way is stored in the variable Z (n) in the work RAM 75 in FIG. 9, and in the next step S16, the timing is synchronized with the sampling clock from the interface 731 in FIG. 8 D
/ A converter 77 is output.

As described in detail above, the human voice signal and the musical tone signal are divided into a plurality of frequency bands as shown in FIG.
The processing of the PF units 90 and 91, and the envelope extraction unit 9 of FIG. 10 for extracting the envelope from the band-limited tone signal.
2, the processing of the multiplying unit 93 in FIG. 10 for applying amplitude modulation to the band-limited audio signal by the envelope signal, and accumulating the modulation output of each band to obtain the output audio signal in FIG. The processing of the unit 94 is realized as digital filter processing by software time division processing.

Thus, the effect adding process of adding the nuance of the musical instrument sound to the overtone component of the human voice can be easily and stably performed by the one-chip DSP. In the embodiment described above, the audio signal and the tone signal are
Although the audio signal is divided into the same frequency band and processed, the audio signal obtained by dividing the audio signal into a certain band is modulated by an envelope signal obtained by dividing the tone signal into another band. Nevertheless, interesting effects can be obtained.

In the operation flowchart of FIG. 22, the sampling frequencies of the audio signal and the tone signal are the same, but when the sampling frequencies are different, each signal is fetched into the memory by interrupt processing, and If the processing is performed at regular intervals, the same effect as in the above-described embodiment can be easily realized. In this case, the band of each bandpass filter is set appropriately.

If there is room in the DSP processing, the bandpass filter arithmetic processing is not performed as the arithmetic processing of the combination of the highpass filter and the lowpass filter as in the above-described embodiment. It may be realized by arithmetic processing configured based on the result of direct design.

[0121]

According to the present invention, for example, when a performer sings a melody or the like in accordance with an accompaniment sound of a music to be automatically played, the voice signal of the singing voice is converted into a musical tone signal of a melody musical instrument, for example. Thus, modulation can be performed for each frequency band.

Therefore, it is possible to add to the timbre of the singing voice a complex and variably changing characteristic according to a specific timbre change of the melody instrument or the like. Further, for example, when the melody instrument is a flute, if a player plays a melody with a whistle instead of a song, it is possible to obtain an effect as if the player were playing the flute freely.

[Brief description of the drawings]

FIG. 1 is an overall configuration diagram of a first embodiment according to the present invention.

FIG. 2 is a block diagram of a modulation circuit 14;

FIG. 3 is a characteristic diagram of a BPF 30/32.

FIG. 4 is a block diagram of another embodiment of the modulation circuit 14;

FIG. 5 is a block diagram of another embodiment of the modulation circuit 14;

FIG. 6 is an overall configuration diagram of a second embodiment according to the present invention.

FIG. 7 is an overall configuration diagram of a third embodiment according to the present invention.

FIG. 8 is an overall configuration diagram of a fourth embodiment according to the present invention.

FIG. 9 is an overall configuration diagram of a DSP.

FIG. 10 is a functional block diagram of a DSP.

FIG. 11 is a filter configuration diagram of a BPF unit and an envelope extraction unit.

FIG. 12 is a configuration diagram of a high-pass filter H ₁ (z).

FIG. 13 is a characteristic diagram of the high-pass filter H ₁ (z).

FIG. 14 is a configuration diagram of a high-pass filter H _2t (z).

FIG. 15 is a diagram showing the relationship between the poles and the zero point of the low-pass filter H _2t (z) and the relationship between the pole vector and the zero vector.

FIG. 16 is an amplitude characteristic diagram of the low-pass filter H _2t (z).

FIG. 17 is a characteristic diagram of the high-pass filter H _2t (z).

FIG. 18 shows bandpass filters H ₁ (z) and H _2t (z).
FIG.

FIG. 19 is a configuration diagram of a low-pass filter H _E (z).

FIG. 20 is a characteristic diagram of a low-pass filter H _E (z).

FIG. 21 is a relationship diagram between | Q _i (n) | and R _j (n).

FIG. 22 is an operation flowchart of a DSP vocoder process.

[Explanation of symbols]

2 tone signal generation circuit 4 ROM pack 6 timbre ROM 8 CPU 10 function switch 12 source 12 _a, 12 _b, 12 _c selection switch 14 modulation circuit 16 the audio signal generating circuit 18 microphone 20, 26 amplifier 22 pronunciation circuit 24, 38 a mixer 28 Speaker 30 BPF 36, 50 VCA 46 Selection circuit 48 Level detection circuit 70 CPU 71 ROM 72 RAM 73 DSP 731 Interface 732 Operation ROM 733 Address counter 734 Decoder 735 Multiplier 736 Adder / Subtractor 737 Register group 738 Flag register 74 Filter coefficient ROM 75 Work RAM 76 tone generator 77 D / A converter 78 keyboard 79 function switch 80 microphone 81, 84 amplifier 82 low-pass filter 83 / D conversion 85 Speaker 86, 87, 88 Bus 90, 91 BPF unit 92, Envelope extraction unit 93 Multiplier 94 Accumulator 102, 103 Delay element 104, 105, 106 Multiplier 107, 108 Adder 109 Absolute value circuit 110 , 111 Delay element 112, 113, 114 Multiplier 115, 116 Adder

Claims (12)

(57) [Claims] A tone input means for inputting 1. A tone signal, a voice input means for inputting a voice signal, the voice input based on the musical tone signal from the tone input means
Modulating means for outputting an external audio signal from the force means as the modulation to output the audio signal, the output level of the tone signal from the tone input means constant Les
Level detection means for detecting that the signal level has fallen below the level, and an audio signal from the audio input means.
The input audio signal is output as the output audio signal.
That the level detection means has fallen below a certain level.
While the audio signal is maintained at a predetermined level.
Level holding means for outputting the output audio signal from the level holding means and the output audio signal from the level holding means and the modulation means.
Mixer means for mixing and outputting the signals . 2. A musical tone input means for inputting a tone signal, a voice input means for inputting an audio signal, the audio signal from the musical tone signal and said audio input hand <br/> stage from said tone input means, respectively Divided into each tone signal and each audio signal band-limited in a plurality of different frequency bands,
Respective modulated based on band-limited each musical tone signals of each audio signal band-limited in the same frequency band in a frequency band, accumulates the respective output which is modulated for each respective frequency band output A modulating means for outputting as an audio signal, and an output level of a tone signal from the tone input means being constant.
Level detection means for detecting that the signal level has fallen below the level, and an audio signal from the audio input means.
The input audio signal is output as the output audio signal.
That the level detection means has fallen below a certain level.
While the audio signal is maintained at a predetermined level.
Level holding means for outputting the output sound signal as the output sound signal, and an output sound signal from the level holding means and the modulation means.
Mixer means for mixing and outputting the signals . Wherein said modulating means comprises a first frequency band dividing means for dividing each tone signal band-limited to a plurality of different frequency bands within the musical tone signals from the tone input means, from said voice input means A second frequency band dividing unit that divides the audio signal into audio signals band-limited to a plurality of different frequency bands the same as the first frequency band dividing unit; An envelope extracting means for extracting each envelope signal from each tone signal; and a characteristic of each audio signal from the second frequency band dividing means being variable by a voltage of each envelope signal from the envelope extracting means corresponding to the same frequency band. Voltage control varying means for accumulating each output from the voltage control varying means, and accumulating means for outputting the output as the output audio signal. 3. The audio modulation device according to claim 2, wherein Wherein said modulating means comprises a first frequency band dividing means for dividing each tone signal band-limited to a plurality of different frequency bands within the musical tone signals from the tone input means, from said voice input means A second frequency band dividing unit that divides the audio signal into audio signals band-limited to the same plurality of different frequency bands as the first frequency band dividing unit; Voltage control variable means for varying the characteristics of each audio signal with the voltage of each tone signal from the first frequency band dividing means corresponding to the same frequency band; and accumulating each output from the voltage control variable means, 3. The audio modulator according to claim 2, further comprising: an accumulating unit that outputs the output audio signal. 5. The voltage control variable unit includes: a voltage of each tone signal from the first frequency band division unit corresponding to the same frequency band, the amplification degree of each audio signal from the second frequency band division unit; The audio modulation device according to claim 3, wherein the audio modulation device is a voltage-controlled amplifier that is variable. Wherein said modulating means is a multiplying means for multiplying a tone signal from the tone input means to the audio signal from the audio input unit, that the voice modulator according to claim 1, wherein the. 7. The music input from the music input means.
A part of a sound signal and the output sound signal from the modulating means;
7. The audio modulation apparatus according to claim 1 , further comprising a mixing unit that mixes and outputs the signals. 8. The musical sound input means includes means for automatically playing music data.
The sound modulation device according to any one of claims 1 to 7, wherein a tone signal generated based on the sound signal is input . 9. The musical sound input means according to claim 1 , wherein said musical sound signal comprises:
The audio modulation device according to claim 1, wherein a melody sound signal and an accompaniment sound signal are input to the modulation unit, and the melody sound signal is input to the modulation unit. 10. The musical sound input means includes: a musical sound signal;
Then, a melody sound signal and an accompaniment sound signal are input, and either the melody signal or the accompaniment signal is selected to change the melody signal or the accompaniment signal.
10. The audio modulation device according to claim 1 , further comprising a selection unit for inputting to the adjustment unit . 11. A tone signal generating means for generating a tone signal.
Voice input means for inputting a voice signal; and the sound based on a tone signal from the tone signal generating means.
Modulates the external audio signal from the voice input means and outputs the output audio signal
Modulating means for outputting as the output level of the tone signal from the tone signal generation means one
Level detection means for detecting when the level falls below a certain level
And an audio signal from the audio input means is input.
The input audio signal is output as the output audio signal.
Is, said level detecting means Tsu name below a predetermined level
While the audio signal is maintained at a predetermined level.
Level holding means for outputting the output audio signal from the level holding means and the output audio signal from the level holding means and the modulation means.
Electrons with a built-in voice modulation apparatus characterized by having, a mixer means for mixing and outputting the
Musical instruments. 12. A tone signal generating means for generating a tone signal.
When a voice input means for inputting a speech signal, music signal and the voice input from the tone signal generation means
The audio signal from the input
Divided into each tone signal and each audio signal whose band is limited within the band
Then, each audio signal band-limited in each frequency band is
On the basis of each of the tone signals band-limited within the same frequency band.
And modulates the output of each frequency band.
Modulating means for outputting calculated to as an output audio signal, the output level of the tone signal from the tone signal generation means one
Level detection means for detecting when the level falls below a certain level
And an audio signal from the audio input means is input.
The input audio signal is output as the output audio signal.
That the level detection means has fallen below a certain level.
While detecting the sound signal, while maintaining the audio signal at a predetermined level.
Level holding means for outputting the output sound signal as the output sound signal, and an output sound signal from the level holding means and the modulation means.
Electrons with a built-in voice modulation apparatus characterized by having, a mixer means for mixing and outputting the
Musical instruments.