JP2016038543A - Effect addition device, method, and program, and electric musical instrument - Google Patents

Abstract

The present invention relates to a technique for adding an effect to a musical sound signal, and adds a distorted sound component that changes variously according to the operation of an operator to the musical sound. An input musical sound signal 207 is input to a filter unit 204, and a predetermined frequency band component is selected from the input musical sound signal 207. The frequency shifter 201 performs frequency shift processing on the musical sound signal output from the filter unit 204. The amplifier 205 performs level control processing corresponding to the state signal 203 on the frequency-shifted tone signal output from the frequency shifter 201. The wet sound signal 209 output from the amplifier 205 is input to the mixing unit 202, mixed with the input musical sound signal 207 whose amplitude is adjusted by the amplifier 206, and output as the output musical sound signal 208. The output musical sound signal 208 includes many harmonic components of the wet sound signal 209 obtained by frequency-shifting the input musical sound signal 207, and can simulate a distortion sound component when the damper pedal is turned off. [Selection] Figure 2

Description

  The present invention relates to an apparatus, a method, a program, and an electronic musical instrument for adding an effect to a musical sound signal.

  In an acoustic piano, when the damper pedal device is released from the state where sound is produced using the damper pedal device, the damper returns to the original position. At this time, it is known that vibration is suppressed when the damper touches the string, but a distortion sound component due to this is generated.

  In a conventional electronic piano, there is known a conventional technique for adding an effect corresponding to an operation of a damper pedal to a musical sound by controlling an amplitude envelope of the musical sound generated according to the operation of the damper pedal device.

  In addition, in an electronic piano, a technique for reproducing a distortion sound component generated according to the operation of a damper mechanism at the time of releasing a key of an acoustic piano by detecting key-off touch information when the key is released and generating a key-off sound is also known. (For example, the technique described in Patent Document 1).

Japanese Patent No. 3587167

  However, in the conventional technique for controlling the amplitude envelope of a musical sound generated according to the operation of the damper pedal device, it is only possible to control the amplitude envelope of the musical sound, and the distortion sound component cannot be reproduced faithfully.

  In the prior art that generates a key-off sound in response to a damper operation, a tone generation channel must be assigned to generate the key-off sound, which consumes the channel. In addition, in the conventional method that simply reads out the key-off sound from the waveform memory and generates it, the distorted sound component is minor, and the distorted sound component that changes variously according to the difference in fine operation of the damper pedal device can be faithfully reproduced. I could not.

  An object of the present invention is to add to a musical tone a distortion sound component that changes variously according to an operation of an operator.

  In one example, the frequency shifter that inputs the original sound signal and executes the frequency shift process, and the level that executes the level control process corresponding to the state signal indicating the operation state of the operation element for the sound signal after the frequency shift process A controller, and a mixer that mixes and outputs the sound signal after the level control process to the original sound signal.

  According to the present invention, it is possible to add a distorted sound component that changes variously according to the operation of the operator to the musical sound.

It is a figure which shows the hardware structural example of embodiment of an electronic keyboard musical instrument. It is a block diagram which shows the structural example of an effect provision part. It is a block diagram which shows the structural example of a frequency shifter. It is a figure explaining the effect of feedback. It is a block diagram which shows the structural example of the analytic signal conversion process part using the Hilbert transform implement | achieved by the FIR filter. It is a figure which shows the example of the amplitude envelope characteristic of a wet sound signal when an operation element is turned off. It is a figure explaining the relationship between the depression amount of an operation element, and the amplitude envelope characteristic of a wet sound signal. It is a figure which shows the example of the amplitude envelope characteristic of a wet sound signal when the musical tone signal based on a new sounding instruction | indication generate | occur | produced immediately after the operator was turned off. It is a block diagram which shows the other structural example of a frequency shifter. It is a flowchart which shows the example of the whole control processing of the electronic keyboard instrument by this embodiment. It is a flowchart which shows the detailed example of a manipulator off process. It is a flowchart which shows the detailed example of a pronunciation process.

  Hereinafter, embodiments for carrying out the present invention will be described in detail with reference to the drawings. FIG. 1 is a diagram illustrating a hardware configuration example of an embodiment of an electronic keyboard instrument. In this embodiment, a CPU (Central Processing Unit) 101, a RAM (Random Access Memory) 102, a ROM (Read Only Memory) 103, an operation element 104, a keyboard 105, and a DSP (Digital Signal Processor) 106 are connected to a bus 110. Are connected to each other. The DSP 106 includes a sound source unit 107 and an effect applying unit 108. An output musical sound signal output from the effect applying unit 108 in the DSP 106 is emitted from a sound system 109 including a D / A (digital-analog converter), an amplifier, a speaker, and the like.

  In accordance with a software program stored in the ROM 103, which is a read-only storage device, the CPU 101 executes computations for generating musical tone signals while exchanging data with the RAM 102, which is a readable / writable storage device, and the computation results are stored in the RAM 102. Can write.

  The CPU 101 can scan the key depression state of the keyboard 105 and the operation state of the operation element 104 which is a damper pedal device, and can issue a sound generation instruction to the sound source unit 107 in the DSP 106 as necessary. The sound source unit 107 reads musical tone waveform data from an internal waveform ROM (not shown) based on a sound generation instruction from the CPU 101, and executes arithmetic processing on the musical tone waveform data.

  A musical tone signal obtained as a calculation result in the sound source unit 107 is input to the effect applying unit 108 in the DSP 106 as an input musical tone signal. The effect applying unit 108 generates and outputs an output musical sound signal to the sound system 109 by executing a wet sound signal generation and mixing process corresponding to the pedal-off operation of the operation element 104 with respect to the input musical sound signal.

  FIG. 2 is a block diagram illustrating a configuration example of the effect applying unit 108 of FIG. 2 includes a frequency shifter 201, a mixing unit 202, a state signal 203, a filter unit 204, amplifiers 205 and 206, an input music signal 207, an output music signal 208, and a wet sound signal 209. The effect applying unit 108 generates a wet sound signal 209 as a sound effect component on the wet side 210 indicated by a broken line frame with respect to the input musical sound signal 207 passing through the dry side 211, and the mixing unit 202 mixes both of them. It has a configuration that can.

  The input musical sound signal 207 output from the sound source unit 107 in FIG. 1 is input to the filter unit 204 on the wet side 210. The wet side 210 includes a filter unit 204, a frequency shifter 201, and an amplifier 205. The filter unit 204 is, for example, a band-pass filter that selects a predetermined frequency band component from the input musical sound signal 207, and is realized by the function of the DSP 106 in FIG. The filter unit 204 controls the filter characteristic fc based on the state signal 203 generated according to the operation state of the operation element 104 in FIG.

  A tone signal composed of a predetermined frequency band component output from the filter unit 204 is input to the frequency shifter 201. The frequency shifter 201 inputs a musical sound signal output from the filter unit 204 and executes frequency shift processing. The frequency shift process will be described later.

  The musical tone signal output from the frequency shifter 201 is input to the amplifier 205. The amplifier 205 executes level control processing according to the state signal 203. This amplifier can be realized by a coefficient multiplier that performs envelope control, which is realized by the DSP 106 of FIG.

  A musical sound signal (hereinafter referred to as “wet sound signal”) 209 output from the amplifier 205 is input to the mixing unit 202. An input musical sound signal 207 whose amplitude is adjusted by an amplifier 206 on the path on the dry side 211 is also input to the mixing unit 202. The mixing unit 202 mixes the wet sound signal 209 whose level is controlled by the amplifier 205 with the input musical sound signal 207, and outputs the mixed result as an output musical sound signal 208.

  Although not explicitly shown in FIG. 2, when a time delay occurs in the frequency shift processing by the frequency shifter 201, a delay device that delays the input musical sound signal 207 in accordance with the time delay may be inserted before the amplifier 206. .

  With the configuration of the effect applying unit 108 in FIG. 2, the input tone signal 207 can be mixed with a wet tone signal 209 that is a harmonic component obtained by frequency shifting a specific frequency band component of the input tone signal 207. . At this time, the state signal 203 is determined in accordance with, for example, the operation state of the operation element 104 which is a damper pedal device, and the gain of the amplifier 205 is controlled by the state signal 203, whereby the input musical sound signal in the mixing unit 202. The mixing ratio of the wet sound signal 209 including the harmonic component with respect to 207 is controlled. Thus, for example, when the damper pedal device as the operation element 104 is depressed, the gain of the amplifier 205 is set to zero via the status signal 203 and the ratio of mixing the wet sound signal 209 with the input musical sound signal 207 is zero. When the depression of the damper pedal device is released, the gain of the amplifier 205 is sharply increased to a predetermined value via the state signal 203, and the gain of the amplifier 205 is reduced from the predetermined value to a predetermined value within a predetermined time after the release. The control is performed to gradually decrease the value until. As a result, immediately after the depression of the damper pedal device is released, the input musical sound signal 207 can be mixed with the wet sound signal 209 containing a large amount of harmonic components in the specific frequency band, and the damper when the acoustic piano is released is released. It becomes possible to reproduce the distortion sound component generated according to the operation of the mechanism. Further, not only the gain of the amplifier 205 but also the filter characteristic of the filter unit 204 and the frequency shift processing characteristic of the frequency shifter 201 are controlled by the state signal 203, so that the harmonic component becomes more complicated according to the operation of the damper pedal device. It is possible to obtain an output musical sound signal 208 in which changes.

  FIG. 3 is a block diagram illustrating a configuration example of the frequency shifter 201 in FIG. The frequency shifter 201 shown in FIG. 3 includes an analytic signal processing unit 301, a real part multiplier 302, an imaginary part multiplier 303, a subtractor 304, a feedback path 305, an amplifier 306, a mixing unit 307, an input musical sound signal 308, a wet A sound signal 309 is included. The frequency shifter 201 in FIG. 3 has a configuration that converts the input musical sound signal 308 output from the filter unit 204 in FIG. 2 into an analysis signal and executes frequency shift processing.

Here, the analysis signal has the original signal which is the input musical sound signal 308 as a real part, and the real signal whose phase is delayed by π / 2 (π is a pi) relative to the original signal as an imaginary part. A complex signal. Now, an actual signal f ₁ [n] having an amplitude A having an angular frequency ω ₀ (ω ₀ > 0) is expressed by the following equation (1). Note that n represents discrete time.

f ₁ [n] = A cos [ω ₀ n] (1)

The real signal f ₂ [n] whose phase is delayed by π / 2 with respect to the real signal of the formula (1) is represented by the following formula (2).

f ₂ [n] = A cos [ω ₀ n−π / 2] = Asin [ω ₀ n] (2)

As a result, the complex signal f [n] having the real signal f ₁ [n] of the expression (1) and the real signal f ₂ [n] of the expression (2) in the real part and the imaginary part, respectively, is expressed by the following expression (3) Indicated by

f [n] = f1 [n] + jf2 [n]
= Acos [ω0 n] + jAsin [ω0 n] (3)

  When the expression (3) is expressed by a complex exponential function using Euler's formula, the following expression (4) is obtained.

f [n] = A {exp [jω ₀ n] + exp [−jω ₀ n]} / 2
+ J [−jA {exp [jω ₀ n] −exp [−jω ₀ n]} / 2]
= Aexp [jω ₀ n] (4)

This complex signal f [n] is defined as an analytic signal. From this, the analysis signal, a special case of complex signals, is defined as a positive signal without the negative frequency component - [omega] ₀ contains only frequency components omega _0.

By using the analysis signal thus obtained, frequency conversion can be performed as follows. For simplicity of explanation, the analysis signal whose angular frequency is ω ₀ ignoring the amplitude A is assumed to be exp [jω ₀ n]. When the angular frequency is converted to ω ₀ + ω ₁ , the analysis signal is expressed by the following equation (5).

exp [j (ω ₀ + ω ₁ ) n] = exp [jω ₀ n] × exp [jω ₁ n]
... (5)

From the equation (5), in order to convert the frequency of the analysis signal exp [jω ₀ n] whose angular frequency is ω ₀ into ω ₀ + ω ₁ , the analysis signal exp [of the original signal whose angular frequency is ω ₀ is used. It can be seen that jω ₀ n] may be simply multiplied by an analysis signal exp [jω ₁ n] having an angular frequency of ω ₁ .

  Since the imaginary part is unnecessary in the calculation for frequency-shifting the musical tone signal, the calculation of only the real part of the equation (5) is the calculation represented by the following equation (6). In the following equation, “Re {}” indicates an operation for extracting a real part, and “Im {}” indicates an operation for extracting an imaginary part.

Re {exp [j (ω ₀ + ω ₁ ) n]}
= Re {exp [jω ₀ n] × exp [jω ₁ n]}
= Re {exp [jω ₀ n]} × Re {exp [jω ₁ n]}
−Im {exp [jω ₀ n]} × Im {exp [jω ₁ n]}
... (6)

From the equation (6), the operation of frequency-shifting the analysis signal exp [jω ₀ n] of the original signal whose angular frequency is ω ₀ by the angular frequency ω ₁ is the real part Re {exp [jω ₀ n of the original analysis signal. ]} Is multiplied by the real part Re {exp [jω ₁ n]} of the reference side analysis signal whose angular frequency is ω ₁ , and the imaginary part Im {exp [jω ₀ n]} of the original analysis signal is angular frequency. It can be understood that this can be realized as an operation of subtracting the result obtained by dividing the imaginary part Im {exp [jω ₁ n]} of the reference-side analytic signal having ω ₁ .

When the original signal is the input musical sound signal 308, the angular frequency component includes not only a single angular frequency ω ₀ component but also a plurality of angular frequency components. On the other hand, the analysis signal on the reference side has a single angular frequency ω ₁ . Therefore, in the actual frequency shift processing, the calculation of the expression (6) is executed for each angular frequency component included in the input musical sound signal 308, and the entire frequency component of the input musical sound signal 308 is frequency shifted by the angular frequency ω _1. Will be.

Here, the real part Re {exp [jω ₁ n]} of the reference side analysis signal can be, for example, a cosine wave signal cos (ω ₁ n), and the imaginary part Im {exp [jω _{1] of the} reference side analysis signal. n]} can be, for example, a sine wave signal sin (ω ₁ n).

  As described above, the frequency shift process can be executed by the functional block surrounded by the broken line frame 310 in FIG. First, the analytic signal processing unit 301 calculates an analytic signal corresponding to the input musical sound signal 308.

Next, the real part multiplier 302 generates a cosine wave signal cos (ω ₁ n) having an angular frequency ω ₁ corresponding to the frequency shift amount with respect to the real part of the analytic signal output from the analytic signal processing unit 301. Multiply.

Similarly, the imaginary part multiplier 303 generates a sine wave signal sin (ω ₁ n) having an angular frequency ω ₁ corresponding to the frequency shift amount with respect to the imaginary part of the analysis signal output from the analytic signal processing unit 301. Multiply.

The cosine wave signal cos (ω ₁ n) and the sine wave signal sin (ω ₁ n) can be obtained as output signals of one oscillator, for example.

  The subtractor 304 subtracts the multiplication result of the imaginary part multiplier 303 from the multiplication result of the real part multiplier 302 and outputs the subtraction result as a wet sound signal 309.

  A frequency shifter 201 shown in FIG. 2 includes a feedback path 305 including an amplifier 306 and a mixing unit 307 in addition to the above-described frequency shift processing calculation function indicated by a broken line frame 310. In this feedback path, the wet sound signal 309 output from the subtractor 304 is fed back to the input side to which the input musical sound signal 308 is input via the amplifier 306 and the mixing unit 307.

  This feedback path 305 makes it possible to further repeatedly perform frequency shift processing on the musical tone signal component subjected to frequency shift processing, and to generate more complex harmonic components. That is, since the frequency shift effect is also obtained for the wet sound signal 309 subjected to frequency shift processing, if the feedback gain is increased by adjusting the gain of the amplifier 306, the harmonic component of the set frequency shift amount Can be easily increased. FIG. 4 is a diagram for explaining the effect of feedback. FIG. 4A is a diagram illustrating an example of frequency characteristics of the output musical sound signal 208 (FIG. 2) without feedback. This frequency characteristic includes a frequency component 300 Hz (Hertz) of the input musical tone signal 308 of the original sound and a frequency component 500 Hz of the wet sound signal 309 obtained by performing a frequency shift process of 200 Hz on the component. . FIG. 4B is a diagram showing an example of frequency characteristics of the output musical sound signal 208 (FIG. 2) when the feedback gain is 50%. It can be seen that this frequency characteristic includes frequencies 700 Hz, 900 Hz, and 1100 Hz, and each frequency component obtained by sequentially shifting the frequency by the set frequency shift amount. FIG. 4B is a diagram illustrating an example of frequency characteristics of the output musical sound signal 208 (FIG. 2) when the feedback gain is about 90%. It can be seen that this frequency characteristic includes more harmonic components. Thus, by providing the feedback path 305, it is possible to obtain the output musical sound signal 208 including the wet sound signal 309 having a complex harmonic component.

  FIG. 5 is a block diagram illustrating a configuration example of the analytic signal processing unit 301 in FIG. 3 using the Hilbert transform realized by an FIR filter.

From the definition of the above-described equations (3) and (4), the analysis signal corresponding to the input tone signal 308 has the input tone signal 308 in the real part and delays the phase of the input tone signal 308 by π / 2. This is a signal having a musical sound signal obtained in this way in the imaginary part. It is known that the process of delaying the phase of the input tone signal 308 by π / 2 can be realized by a Hilbert transformer. The frequency characteristic H (ω) of the Hilbert transformer has the following characteristic (7) when the sampling frequency is ω _s .

H (ω) = − j, 0 <ω <ω _s / 2
= J, −ω _s / 2 <ω <0 (7)

  Thus, the Hilbert transformer can be regarded as a filter whose amplitude characteristic is constant regardless of frequency, and whose phase characteristic is delayed by π / 2 in the positive frequency region and advanced by π / 2 in the negative frequency region. Then, by limiting the frequency band of the input musical sound signal 308 by the filter unit 204 of FIG. 2, the filter of the Hilbert transformer having the frequency characteristic of the equation (7) is, for example, an FIR (Finite Impulse Response) filter. Can be calculated approximately with sufficient accuracy.

  Here, when considering real-time processing for a musical sound signal, an M / 2 delay occurs in the Mth-order FIR filter. Therefore, when the Mth-order FIR filter processing is performed on the input musical sound signal 308, the input musical sound analysis signal imaginary number part 502, which is the imaginary part of the analysis signal corresponding to the input musical sound signal 308 output therefrom, is also a real number. The input musical sound analysis signal real number part 501 (= input musical sound signal 308) is a delay of M / 2. Therefore, in order to correct this phase shift, it is necessary to input the input musical sound signal 308 to a delay unit that performs M / 2 delay processing, and output the input musical sound analysis signal real number part 501 as its output.

FIG. 5 is a configuration example of an M-th order FIR filter that realizes the analytic signal processing unit 301 of FIG. In this FIR filter, the input musical sound signal 308 x [n] is input to a circuit portion in which M discrete time unit delay elements indicated by z ⁻¹ are cascade-connected. Then, each filter coefficient in the multiplier to an output of the delay elements _{h 0, h 1, ··· h} M / 2, ··· h M-1, adds the multiplication result obtained by multiplying the h _M The result is output to the imaginary part multiplier 303 of FIG. 3 as y _Im [n] = input musical sound analysis signal imaginary part 502.

In the configuration of FIG. 5, the signal appearing at the output of the delay element at the M / 2 stage of the FIR filter is the signal x [n−M / 2] obtained by delaying the input music signal 308 = x [n] by M / 2. The output x [n−M / 2] of the delay element of the M / 2 stage of the FIR filter is expressed as y _Re [n] = input real sound analysis signal real part without preparing a delay device. 501 can be output to the real part multiplier 302 of FIG.

  The effect providing unit 108 of FIG. 1 having the configuration examples of FIGS. 2 to 5 described above can be realized by software processing of the DSP 106 of FIG.

  Next, the operation when the state signal 203 in FIG. 1 is controlled based on the operation state of the operation element 104 in FIG. 1 will be described.

First, a case where the operation state of the operation element 104 takes a binary value of on or off, for example, will be described. FIG. 6 is a diagram showing an example of the amplitude envelope characteristic of the wet sound signal 209 of FIG. 2 when the operation element 104 is turned off. Predetermined values are set in advance for the filter coefficients h _{0 to} h _M of the FIR filter of FIG. 5, the frequency shift amount determined by the angular frequency ω ₁ of FIG. 3, and the feedback gain set in the amplifier 306 of FIG. Keep it. When the off of the operation element 104 is detected, the state signal 203 in FIG. 2 is controlled as a gain value of the amplifier 205 that controls the amplitude envelope of the wet sound signal 209 in accordance with, for example, a predetermined envelope illustrated in FIG. . That is, the value of the state signal 203 given as the gain of the amplifier 205 is set to zero when the damper pedal device that is the operation element 104 is depressed, and the state signal 203 is decremented when the depression of the damper pedal device is released. After the value is sharply increased to a predetermined value, the value of the state signal 203 is gradually decreased from the predetermined value to zero within a predetermined time after the release. In the example of FIG. 6, the envelope characteristic has a linear transition, but is not limited to this. In this manner, the output tone signal 208 in which the wet tone signal 209 including the harmonic component is added to the input tone signal 207 in FIG.

  Next, a case where the operation state of the operation element 104 takes a value of two or more will be described. In this case, the state signal 203 of FIG. 2 can be controlled based on the start position information when the operation element 104 is turned off and the displacement speed (corresponding to velocity) at the time of turning off. As the operation state of the operation element 104 in this case, the depression amount and depression release speed of the damper pedal device when the depression of the damper pedal device is released are detected. FIG. 7 is a diagram for explaining the relationship between the depression amount of the operation element 104 and the amplitude envelope characteristic of the wet sound signal 209 in FIG. The amplitude envelope characteristic 701 is when the amount of depression of the operation element 104 is large, and the amplitude envelope characteristic 702 is when the amount of depression of the operation element 104 is small. In this way, the maximum level of the value of the state signal 203 supplied to the amplifier 205 is controlled based on the depression amount of the operation element 104, so that the operation element 104 can be controlled according to the depression amount of the operation element 104. When off, the ratio of the wet sound signal 209 in FIG. 2 included in the output musical sound signal 208, that is, the ratio of harmonics can be controlled.

  Next, the depression release speed of the operation element 104 can correspond to the degree of string vibration suppression in the keyboard instrument, and the effect can be increased as the speed increases. Here, in order to reproduce a state where the effect of the degree of suppression is large, the following control can be performed. That is, when the depressing speed is high, the state signal 203 that widens the selection band of the filter unit 204 in FIG. 2 is supplied to the filter unit 204, and the feedback gain of the amplifier 306 in FIG. 3 is increased. The signal 203 is supplied to the amplifier 306. When the stepping-off release speed is slow, a state signal 203 that narrows the selection band of the filter unit 204 in FIG. 2 is supplied to the filter unit 204, and a state signal 203 that lowers the feedback gain of the amplifier 306 in FIG. Is supplied to the amplifier 306.

  By controlling the depression amount and depression release speed of the operation element 104 as described above, for example, when the damper pedal is released quickly while the depression of the operation element 104 is large, the wet sound signal 209 has many harmonic components. The output musical sound signal 208 can be output in a state of being included.

  Here, the frequency shift amount is fixed, but it is also possible to adjust the frequency shift amount and the feedback gain as appropriate to adjust the density of the harmonics more finely.

  Next, in the analysis signal output by the analysis signal conversion processing unit 301 in FIGS. 3 and 5, the envelope can be generally calculated by the following equation (8).

(Y _Re [n] ² + y _Im [n] ² ) ^1/2 (8)

  By using the calculated value of equation (8) or the value before taking the square root, it is possible to know the approximate amplitude (approximately level information) of the analysis signal. When the state signal 203 is determined when the operation element 104 is turned off, this approximate amplitude information can also be taken into consideration. For example, when the value of the approximate amplitude is small, it is considered that the influence on the control target at that timing is also small, and the degree of influence on each parameter by the state signal 203 may be small. If it is effective, it is sufficient to use a product obtained by multiplying the ratio by the approximate amplitude ratio. As a result, it is possible to change the degree of application of the effect according to the amplitude of the musical sound signal that is actually sounding. For example, when the effect is applied, if the input musical sound signal 207 has a small amplitude, it is possible to eliminate such unnaturalness that the ratio of the wet sound signal 209 in FIG. 2 becomes extremely large.

  FIG. 8 is a diagram illustrating an example of the amplitude envelope characteristic of the wet sound signal 209 when the input musical sound signal 207 based on a new sound generation instruction is generated from the sound source unit 107 of FIG. 1 immediately after the operator 104 is turned off. . For example, when a new input musical sound signal 207 is generated at the timing of 801 in FIG. 8 during the generation process of the wet sound signal 209 for the operation element 104 being turned off, the string of the keyboard instrument is again in a vibration state. Therefore, the processing for reducing the influence of the wet sound signal 209 is executed. Specifically, when a new sound is detected during the generation process of the wet sound signal 209, an envelope in which the amplitude envelope of the wet sound signal 209 is muted faster than the normal envelope 802 as shown in FIG. It is switched to 803.

FIG. 9 is a block diagram illustrating another configuration example of the frequency shifter 201 in FIG. The configuration of FIG. 9 differs from the configuration of FIG. 3 in that the cosine wave signal cos (ω ₁ n) having an angular frequency ω ₁ corresponding to the frequency shift amount supplied to the real part multiplier 302 and the imaginary part multiplier A sinusoidal wave signal sin (ω ₁ n) having the same angular frequency ω ₁ supplied to 303 is supplied not from the oscillator but from the sound source unit 107 in FIG. Since the sound source unit 107 generally has a function of generating these signals, it is not necessary to separately prepare an oscillator, and the configuration of the effect applying unit 108 can be simplified. When this configuration is adopted, it is possible to reduce the amount of processing in the effect imparting unit 108 while utilizing existing resources, and at the same time, the amount of shift and the like can be more finely controlled on the sound source unit 107 side.

  FIG. 10 is a flowchart showing an example of the overall control process of the electronic keyboard instrument according to the present embodiment. This overall processing is realized as an operation in which the CPU 101 in FIG. 1 executes the overall control processing program stored in the ROM 103 while using the RAM 102 as a work area.

  When the power switch (not shown) of the electronic keyboard instrument is turned on by the user, the CPU 101 starts this program, and until the power switch is turned off, the operator off process in step S1001 and the sound generation process in step S1002 , And other processes in step S1003 are repeatedly executed. The other processing in step S1003 is processing for scanning various setting switches (not shown) of the electronic keyboard instrument.

  FIG. 11 is a flowchart showing a detailed example of the operator off process in step S1001 of FIG.

  First, the CPU 101 obtains time position information (depression amount) when starting to turn off the operation element 104 and stores it in a variable x on the RAM 102 (step S1101).

  Next, the CPU 101 acquires the input amplitude information of the input musical sound signal 207 of FIG. 2 by the above-described equation (8) and stores it in the variable env on the RAM 102 (step S1102).

  Next, the CPU 101 obtains a Wet level that is a value of the state signal 203 (gain supplied to the amplifier 205) for determining the amplitude of the wet sound signal 209 in FIG. 2 (step S1103). Specifically, a function LevelCurve (x, env) for calculating a Wet level using the time position information of the off process start acquired in the variable x in step S1101 and the input amplitude information acquired in the variable env in step S1102 as parameters. The calculation result is stored in a variable wet on the RAM 102.

  Next, the CPU 101 acquires the depression release speed information of the operation element 104 and stores it in the variable v on the RAM 102 (step S1104).

Next, the CPU 101 uses a function Coef (v) that calculates the center frequency of the filter unit 204 in FIG. 2 using the stepping release speed information acquired in the variable v in step S1104 and the input amplitude information acquired in the variable env in step S1102 as parameters. , Env) and the calculation result is stored in a variable f _c on the RAM 102. Further, the CPU 101 calculates a frequency shift amount (corresponding to ω ₁ in FIG. 3) using the stepping release speed information acquired in the variable v in step S1104 and the input amplitude information acquired in the variable env in step S1102 as parameters. (V, env) is called and the calculation result is stored in a variable shift on the RAM 102. Further, the CPU 101 uses the depression release speed information acquired in the variable v in step S1104 and the input amplitude information acquired in the variable env in step S1102 as parameters, and the value of the state signal 203 supplied as a feedback gain to the amplifier 306 in FIG. The function Feedback (v, env) to be calculated is called, and the calculation result is stored in the variable fbk on the RAM 102 (step S1105).

Thereafter, the CPU 101 notifies the effect imparting unit 108 in the DSP 106 of FIG. 1 of the parameters acquired in the variables f _c , shift, and fbk in step S1105 (step S1106). Thereby, the DSP 106 sets the center frequency of the filter unit 204 in FIG. 2, the frequency shift amount ω ₁ in FIG. 3, and the gain of the amplifier 306 in FIG. 3 based on the values of the variables f _c , shift, and fbk. Control.

  Further, the CPU 101 notifies the effect acquisition unit 108 in the DSP 106 of FIG. 1 of the parameter acquired in the variable wet in step S1103 (step S1107). As a result, the DSP 106 controls the gain of the amplifier 205 in FIG. 2 based on the value of the variable wet.

  After the series of processes described above, the CPU 101 ends the operator-off process in step S1001 of FIG. 10 illustrated by the flowchart of FIG.

  In the flowchart of FIG. 11, the processes in steps S1102, S1104, and S1105 surrounded by a broken line are performed when the operation information of the operator 104 takes a binary value of on and off as described in FIG. Does not have to run. In this case, as described above, only the gain of the amplifier 205 in FIG. In this case, the value of the variable env referred to in step S1103 is a fixed value.

  FIG. 12 is a flowchart showing a detailed example of the sound generation process in step S1002 of FIG.

  First, the CPU 101 scans the state of the keyboard 105 to detect key depression or key release, and notifies the sound source unit 107 of a key depression (note on) instruction or key release (note off) instruction based on the detection result. A normal sound generation process such as the above is executed (step S1201).

  Next, the CPU 101 determines whether or not the effect applying unit 108 in the DSP 106 in FIG. 1 is executing envelope control (step S1202).

  When the effect imparting unit 108 is not executing the envelope control (when the determination in step S1202 is NO), the CPU 101 ends the sound generation process in step S1002 of FIG. 10 illustrated in the flowchart of FIG. 12 as it is.

  When the effect imparting unit 108 is executing envelope control (when the determination in step S1202 is YES), the CPU 101 executes processing for setting an envelope value for high-speed release as illustrated as 803 in FIG. The value obtained as a result is stored in the variable wet on the RAM 102 (step S1203).

  The CPU 101 notifies the effect imparting unit 108 in the DSP 106 of FIG. 1 of the parameter acquired in the variable wet in step S1203 (step S1204). As a result, the DSP 106 performs high-speed release processing on the gain of the amplifier 205 in FIG. 2 with the characteristic illustrated by 803 in FIG. 8 based on the value of the variable wet.

  The embodiment described above is configured to perform frequency shift processing on the entire input musical sound signal 207 output from the sound source unit 107 of FIG. 1, but it is possible to input individual channels output from the sound source unit 107. A frequency shift process may be performed on the musical sound signal.

  According to the embodiment described above, the harmonic structure by the frequency shift process can be changed according to the input musical sound signal 207, so that the effect is not monotonous as in the case of adding a distortion sound component by the conventional waveform reproduction, and the damper You can feel the distortion when you take your foot off the pedal device.

  In the present embodiment, by providing the filter unit 204 of FIG. 2, it is possible to generate a frequency-shifted sound based on a sound in a specific band, so that the characteristics of the distorted sound can be easily controlled.

  In the present embodiment, since the feedback path 305 of FIG. 3 is provided in the frequency shifter 201 of FIG. 2, it is possible to adjust the frequency component of the generated signal in addition to the adjustment of the frequency shift amount, and complex harmonics. It becomes possible to add components.

  In the present embodiment, since the frequency shift process can be controlled based on the amplitude information of the input musical sound signal 207, the effect does not become monotonous but becomes a natural movement.

  The system according to the present embodiment is realized by the CPU 101 in FIG. 1 executing a program having functions implemented by the processing of the flowcharts in FIGS. For example, the program may be recorded and distributed on an external storage device or a portable recording medium not specifically shown in FIG. 1, or may be acquired from a network by a communication interface not particularly shown in FIG. Also good. Also, the control program executed by the DSP 106 in FIG. 1 can be acquired in the same manner.

Regarding the above embodiment, the following additional notes are disclosed.
(Appendix 1)
A frequency shifter that inputs an original sound signal and executes frequency shift processing;
A level controller that executes a level control process corresponding to a state signal representing an operation state of the operator with respect to the sound signal after the frequency shift process;
A mixer that mixes and outputs the sound signal after the level control processing to the original sound signal;
An effect adding device comprising:
(Appendix 2)
A filter for selecting a predetermined frequency band component from the original sound signal, and inputting the sound signal of the predetermined frequency band component to the frequency shifter;
The effect adding device according to supplementary note 1, wherein:
(Appendix 3)
The filter controls a filter characteristic according to the state signal.
The effect adding device according to Supplementary Note 2, wherein
(Appendix 4)
The frequency shifter includes a feedback path that feeds back a sound signal output from the frequency shifter to the input side of the frequency shifter.
The effect adding device according to any one of appendices 1 to 3, characterized in that:
(Appendix 5)
The frequency shifter controls a feedback amount in the feedback path according to the state signal.
The effect adding device according to appendix 4, characterized in that:
(Appendix 6)
The frequency shifter is
Analysis signal conversion processing means for inputting the original sound signal and calculating an analysis signal;
Multiplication means for multiplying the real part and imaginary part of the analytic signal respectively by the real part and imaginary part of the analytic signal having a frequency corresponding to the frequency shift amount in the frequency shift processing,
Adding means for adding the real part and the imaginary part output by the multiplication means, and outputting the sound signal after the addition as the sound signal after the frequency shift processing;
The effect adding device according to any one of appendices 1 to 5, further comprising:
(Appendix 7)
An analysis signal having a frequency corresponding to the frequency shift amount is generated based on the output of the sound source unit of the electronic musical instrument.
The effect adding device according to appendix 6, characterized in that:
(Appendix 8)
Generating approximate level information of the analysis signal output by the analytic signal processing means, and generating the state signal based on the approximate level information;
The effect adding device according to appendix 6 or 7, characterized in that:
(Appendix 9)
The effect adding device according to any one of appendices 1 to 8,
A sound source means for generating a musical sound signal;
With
The musical sound signal is input as the original sound signal to the effect adding device, and the sound signal output by the effect adding device is output as an output musical sound signal.
An electronic musical instrument characterized by that.
(Appendix 10)
The operation element is a damper pedal device, and the operation state of the operation element is a state in which the depression of the damper pedal device is released.
The electronic musical instrument according to appendix 9, characterized in that.
(Appendix 11)
The state signal representing the state of the operation element is a signal representing the depression amount and depression release speed of the damper pedal device when the depression of the damper pedal device is released.
The electronic musical instrument according to appendix 10, wherein the electronic musical instrument is characterized in that
(Appendix 12)
The mixer sets the ratio of mixing the sound signal after the level control processing to the original sound signal when the damper pedal apparatus is depressed, and the ratio when the depression of the damper pedal apparatus is released. Is sharply increased to a predetermined ratio, and then the ratio is gradually decreased from the predetermined ratio to zero within a predetermined time after the time of the release,
The electronic musical instrument according to any one of appendices 9 to 11, characterized in that.
(Appendix 13)
The mixer controls the predetermined ratio in accordance with a depression amount of the damper pedal device;
The electronic musical instrument according to appendix 12, characterized in that.
(Appendix 14)
The mixer performs the predetermined decrease when the musical tone signal based on the next sound generation instruction is generated from the sound source means while executing the control for gradually decreasing the ratio from the predetermined ratio to zero within the predetermined time. Gradually decreasing the rate from the predetermined rate to zero within a time shorter than the time;
The electronic musical instrument according to appendix 12 or 13, characterized by the above.
(Appendix 15)
Effect adding device
Input the original sound signal and execute frequency shift processing,
A level control process corresponding to the state signal including the operation state of the operator is performed on the sound signal after the frequency shift process,
The sound signal after the level control processing is mixed with the original sound signal and output.
An effect adding method characterized by that.
(Appendix 16)
Inputting an original sound signal and performing frequency shift processing;
Performing a level control process according to a state signal including an operation state of an operator on a sound signal after the frequency shift process;
Mixing and outputting the sound signal after the level control processing to the original sound signal;
A program that causes a computer to execute.

101 CPU
102 RAM
103 ROM
104 Controller 105 Keyboard 106 DSP
DESCRIPTION OF SYMBOLS 107 Sound source part 108 Effect provision part 109 Sound system 110 Bus 201 Frequency shifter 202,307 Mixing part 203 Status signal 204 Filter part 205,206,306 Amplifier 207,308 Input musical sound signal 208 Output musical sound signal 209,309 Wet sound signal 210 Wet Side 211 Dry side 301 Analysis signal conversion processing unit 302 Real number part multiplier 303 Imaginary number part multiplier 304 Subtractor 305 Feedback path 310 Broken line frame 501 Input musical sound analysis signal real part 502 Input musical sound analysis signal imaginary part

Claims (16)

A frequency shifter that inputs an original sound signal and executes frequency shift processing;
A level controller that executes a level control process corresponding to a state signal representing an operation state of the operator with respect to the sound signal after the frequency shift process;
A mixer that mixes and outputs the sound signal after the level control processing to the original sound signal;
An effect adding device comprising: A filter for selecting a predetermined frequency band component from the original sound signal, and inputting the sound signal of the predetermined frequency band component to the frequency shifter;
The effect adding device according to claim 1. The filter controls a filter characteristic according to the state signal.
The effect adding device according to claim 2. The frequency shifter includes a feedback path that feeds back a sound signal output from the frequency shifter to the input side of the frequency shifter.
The effect adding device according to any one of claims 1 to 3, wherein The frequency shifter controls a feedback amount in the feedback path according to the state signal.
The effect adding device according to claim 4, wherein: The frequency shifter is
Analysis signal conversion processing means for inputting the original sound signal and calculating an analysis signal;
Multiplication means for multiplying the real part and imaginary part of the analytic signal respectively by the real part and imaginary part of the analytic signal having a frequency corresponding to the frequency shift amount in the frequency shift processing,
Adding means for adding the real part and the imaginary part output by the multiplication means, and outputting the sound signal after the addition as the sound signal after the frequency shift processing;
The effect adding device according to any one of claims 1 to 5, further comprising: An analysis signal having a frequency corresponding to the frequency shift amount is generated based on the output of the sound source unit of the electronic musical instrument.
The effect adding device according to claim 6. Generating approximate level information of the analysis signal output by the analytic signal processing means, and generating the state signal based on the approximate level information;
The effect adding apparatus according to claim 6 or 7, wherein The effect adding device according to any one of claims 1 to 8,
A sound source means for generating a musical sound signal;
With
The musical sound signal is input as the original sound signal to the effect adding device, and the sound signal output by the effect adding device is output as an output musical sound signal.
An electronic musical instrument characterized by that. The operation element is a damper pedal device, and the operation state of the operation element is a state in which the depression of the damper pedal device is released.
The electronic musical instrument according to claim 9. The state signal representing the operation state of the operation element is a signal representing the depression amount and depression release speed of the damper pedal device when the depression of the damper pedal device is released.
The electronic musical instrument according to claim 10. The mixer sets the ratio of mixing the sound signal after the level control process to the original sound signal when the damper pedal apparatus is depressed, and the ratio when the depression of the damper pedal apparatus is released. Is sharply increased to a predetermined ratio, and then the ratio is gradually decreased from the predetermined ratio to zero within a predetermined time after the time of the release,
The electronic musical instrument according to claim 9, wherein the electronic musical instrument is any one of the electronic musical instruments. The mixer controls the predetermined ratio in accordance with a depression amount of the damper pedal device;
The electronic musical instrument according to claim 12, wherein: The mixer performs the predetermined decrease when the musical tone signal based on the next sound generation instruction is generated from the sound source means while executing the control for gradually decreasing the ratio from the predetermined ratio to zero within the predetermined time. Gradually decreasing the rate from the predetermined rate to zero within a time shorter than the time;
The electronic musical instrument according to claim 12 or 13, characterized by the above. Effect adding device
Input the original sound signal and execute frequency shift processing,
A level control process corresponding to the state signal including the operation state of the operator is performed on the sound signal after the frequency shift process,
The sound signal after the level control processing is mixed with the original sound signal and output.
An effect adding method characterized by that. Inputting an original sound signal and performing frequency shift processing;
Performing a level control process according to a state signal including an operation state of an operator on a sound signal after the frequency shift process;
Mixing and outputting the sound signal after the level control processing to the original sound signal;
A program that causes a computer to execute.