JP4655811B2 - Acoustic signal processing apparatus and program - Google Patents

Description

  The present invention provides an acoustic signal processing device having an effect function of mixing and outputting an input signal and a signal obtained by performing processing on the input signal, and for causing a computer to function as such an acoustic signal processing device. Related to the program.

Conventionally, an effector for adding an effect called a doubling effect that mixes and outputs an input signal and a signal called a voice signal obtained by processing the input signal is known.
In such an effector, the voice signal is generated by delaying the input signal using a delay, or by performing a pitch shift that simply shifts the pitch of the input signal by a certain amount. .
The output signal is obtained by superimposing the input signal and the voice signal, or panning these signals left and right by panning.

Such effectors are described in, for example, Patent Documents 1 to 5 below.
Japanese Patent No. 3183117 US Pat. No. 5,301,259 US Pat. No. 5,231,671 US Pat. No. 5,567,901 US Pat. No. 6,046,395

However, in the conventional effector, the voice signal is generated based on the same reference regardless of the pitch of the input signal. For this reason, even if the pitch of the input signal deviates from the scale sound, the difference is emphasized, or the pitch of the voice signal deviates greatly from the pitch of the scale sound. On the other hand, there was a problem that an unnatural hearing might be caused.
An object of the present invention is to solve such a problem and realize an effect capable of always obtaining a natural audible output signal.

In order to achieve the above object, an acoustic signal processing device according to the present invention includes a processing signal generating means for generating a processing signal by subjecting an input signal to pitch conversion processing, and a mixture of the input signal and the processing signal for output. and mixing means for the pitch storage means for storing the pitch of all the chromatic notes in a predetermined range of a scale, provided with estimation means for estimating a chromatic note corresponding to the input signal, the processing signal generating means, the If the estimating means is able to estimate the chromatic note corresponding to the input signal only, and means for generating the processing signal I row the pitch conversion processing to approximate the pitch of the input signal to the pitch of the estimated chromatic note all SANYO that, the estimation means, and the pitch of the input signal, when the difference between the pitch of a scale notes stored in said pitch storage means within a predetermined error range, the chromatic note is the It is a means for estimating that the scale signal corresponds to the force signal, and the predetermined error range is set for all the scale sounds for which the pitch storage means stores the pitch, and the error range for a certain tone is set for other scale sounds. It is determined so as not to overlap with the error range for the scale sound.

  In addition, the present invention can be configured and implemented as an apparatus invention, and is also implemented in the form of a method, a program of a computer or a processor such as a digital signal processor, and a storage medium storing such a program. You can also.

According to the acoustic signal processing apparatus of the present invention as described above, it is possible to realize an effect that can always obtain a natural audible output signal.
Moreover, according to the program of this invention, a computer can be functioned as an acoustic signal processing apparatus, and the same effect can be acquired.

Hereinafter, the best mode for carrying out the present invention will be specifically described with reference to the drawings.
[First Embodiment: FIGS. 1 to 16]
First, the configuration of an electronic musical instrument which is a first embodiment of the acoustic signal processing apparatus of the present invention will be described with reference to FIG. FIG. 1 is a block diagram showing the configuration of the electronic musical instrument.
As shown in FIG. 1, the electronic musical instrument 10 includes a CPU 11, a ROM 12, a RAM 13, a detection circuit 14, a display circuit 15, an audio signal interface (I / F) 16, a communication I / F 17, a sound source unit 18, and a signal processing unit 19. These are connected by a system bus 20. An operation element 21 is connected to the detection circuit 14, a display 22 is connected to the display circuit 15, and a sound system 23 is connected to the signal processing unit 19.

  The CPU 11 is a control unit that performs overall control of the electronic musical instrument 10, and by executing a required control program stored in the ROM 12, the operation content detection and display circuit 15 of the operation element 21 via the detection circuit 14 is detected. Display control of the display device 22 via the audio signal I / F 16, input reception of the audio signal via the audio signal I / F 16, control of communication via the communication I / F 17, control of waveform data generation in the sound source unit 18, signal in the signal processing unit 19 Control operations such as processing control are performed.

The ROM 12 is a storage unit that stores a control program executed by the CPU 11, data that does not need to be changed, and the like. It is conceivable that the ROM 12 is constituted by rewritable nonvolatile storage means such as a flash memory so that these data can be updated.
The RAM 13 is a storage unit that is used as a work memory for the CPU 11 or that temporarily stores parameter values and the like.

  The detection circuit 14 is a circuit for detecting an operation content performed on the operation element 21 and transmitting a signal according to the content to the CPU 11. The operation element 21 is constituted by keys, buttons, dials, sliders, and the like, and is an operation means for receiving an operation from the user on the electronic musical instrument 10. Note that the display 22 and the operation element 21 can be integrally formed by, for example, stacking a touch panel on the LCD. In addition, depending on the type of the electronic musical instrument 10, an operation element for receiving a performance operation such as a keyboard, strings, pads, pedals, and a breath controller is included.

  The display circuit 15 is a circuit that controls display on the display 22 in accordance with an instruction from the CPU 11. The display 22 is configured by a liquid crystal display (LCD), a light emitting diode (LED) lamp, or the like, and is a graphical user for receiving the operating state and setting contents of the electronic musical instrument 10, a message to the user, and an instruction from the user. Display means for displaying an interface (GUI) or the like.

  The audio signal I / F 16 is an interface for connecting a microphone and other audio equipment and receiving an input of an audio signal. The audio signal input here is used for signal processing in the signal processing unit 19. At this time, the signal may be buffered by the RAM 13 or the like. When receiving an analog signal input, A / D conversion is performed to convert it into a digital signal.

The communication I / F 17 is an interface for communicating with an external device such as a PC (personal computer) by connecting to a network such as a LAN (local area network). For example, an Ethernet (registered trademark) standard interface can be used.
Further, as the communication I / F 17, an interface for transmitting / receiving MIDI data to / from an external device handling MIDI data, such as another electronic musical instrument or a sound source device, may be provided. Such an interface can be configured by an interface conforming to, for example, the USB standard, the IEEE 1394 (Institute of Electrical and Electronic Engineers 1394) standard, or the RS232C (Recommended Standard 232 version C) standard. It may be possible to transmit and receive MIDI data and other data via a common interface.

  The sound source unit 18 is a digital sound signal in a plurality of sound generation channels based on MIDI performance data generated by the CPU 11 according to the operation of the performance operator or received from an external device via the communication I / F 17. This is a sound source means for generating waveform data. The generated waveform data is input to the signal processing unit 19 for signal processing.

The signal processing unit 19 functions as an effector, a mixer, and the like, and applies effects according to processing parameters set by the CPU 11 to waveform data generated by the sound source unit 18 or input via the audio signal I / F 16. And signal processing means for performing signal processing such as mixing. Further, the processed signal is input to the sound system 23, and sound generation based on the signal is performed.
The sound source unit 18 and the signal processing unit 19 may be realized by software or hardware.

  By the way, the electronic musical instrument 10 described above includes a doubling effector that applies a doubling effect to the input signal in the signal processing unit 19. Here, doubling refers to a process of adding a processed sound to the original sound to increase the thickness of the sound. Such doubling itself is widely used as an effect simulating double track recording or ADT (Artificial Double Tracking), and is often used not only for vocals but also for musical instruments such as electric guitars.

In addition, as the above-mentioned “processing”, for example, delay, panning, pitch shift, or a process in which these are combined with modulation or detune has been used.
The “processing” performed in the electronic musical instrument 10 can also be used as the contents of the processing, but these “processing” are performed in consideration of the relationship between the pitch of the input sound and the pitch of the scale sound. This is a feature of this embodiment.

Next, FIG. 2 shows a functional configuration of the doubling effector as described above provided in the signal processing unit 19.
As shown in FIG. 2, the doubling effector 30 includes a voice signal generation unit 40, a delay processing unit 50, and a mixing unit 60.
Among these, the voice signal generation unit 40 is a processing signal generation means for generating a voice signal by performing a pitch conversion process on the input signal, and its configuration is shown in FIG.
As shown in FIG. 3, the voice signal generation unit 40 includes a pitch detection unit 41, a pitch buffer 42, a scale sound estimation unit 43, a pitch table 44, an estimated value buffer 45, a pitch processing unit 46, and a pitch conversion unit 47. It has.

The pitch detector 41 is a pitch detector having a function of acquiring pitch information by detecting the pitch (time length of one cycle or its frequency) of the input signal. Details of the pitch detection process for acquiring pitch information will be described later.
The pitch buffer 42 has a function of temporarily storing a sample of the pitch of the input signal detected by the pitch detector 41. The pitch buffer may be a ring buffer that erases old data when the capacity is full.

  The scale sound estimation unit 43 is an estimation unit having a function of estimating a scale sound corresponding to an input signal. Then, the pitch of the input signal stored in the pitch buffer 42 is compared with the pitch of the scale sound stored in the pitch table 44, and the difference between the pitch of the input signal and the pitch of a certain scale sound is predetermined. If it is within the error range, it is estimated that the scale sound corresponding to the input signal is the scale sound.

More specifically, for example, a pitch Scale (n) = 440 × 2 ^{(n−69) / 12} Hz (Hertz) is applied to the pitch table 44 as pitches of equal temperament, with pitches n = 0 to 127. In addition, the allowable error range Δh (n) of each pitch is set to an appropriate value, and n satisfying the following formula 1 is present for the pitch P _I (frequency in this case) of the input signal. It is conceivable to estimate that the pitch of the scale sound corresponding to the input signal is n.

In this case, the value of Δh (n) may be different depending on n, but is determined so as to satisfy the following formula 2. That is, it is determined so that the allowable error ranges related to the scale sounds do not overlap. Also, if P _I do not even meet the number 1 for any n, it estimates that the input signal does not correspond to any scale sound, so as to output an invalid value as the estimated value.

Further, at a certain time t, only if P _I for the same n continuously from [Delta] w (n) by a past time (i.e., time t-Δw (n)) to the current satisfies the above Equation 1, corresponding to the input signal You may make it estimate that the pitch of a scale sound is n. While the input signal shifts from one scale sound to another, it is possible that the scale sound during that time will satisfy Equation 1 for a very short time. In such a case, the input signal is This is because it is more appropriate not to think that it is compatible.

Here, this method is adopted, and the scale sound estimation unit 43 stores the pitch of the scale sound estimated by Equation 1 in the estimated value buffer 45, and the scale sound estimated by Equation 1 is Δw (n). The pitch stored in the estimated value buffer 45 in the pitch processing unit 46 as the estimated scale pitch P _E that is the pitch of the scale sound estimated to correspond to the input signal. I am trying to output. And when there is no chromatic note is estimated by the number 1, if not by the same pitch between the [Delta] w (n) is them to data to that effect as the estimated chromatic note pitch P _E to output the pitch processing unit 46 That's fine. Note that the value of Δw (n) may be different depending on n.

FIG. 4 shows an example of the estimation result when the scale sound corresponding to the input signal is estimated by the above method.
As shown in this figure, when the pitch of the input signal changes as shown by the solid line 71 with the passage of time, if the pitch of the input signal is estimated by the above-described method, the input signal is It is estimated that the pitch of the corresponding scale sound is k, and the pitch of the scale sound corresponding to the input signal is k-1 between the times t5 and t6. The tolerance dashed 72-75 is the pitch of each note in the scale, thick lines 76 and 77 shows a chromatic note and its pitch corresponding to the input signal P _I. Before time t1 to be, although _{P I} is within the allowable error range of Scale (k-1), the time is shorter than Δw (k-1), there is no chromatic note corresponding to _{P I} here It is estimated that.

Incidentally, when performing effect processing in real time, but do not know the pitch P _I of an input signal at a future time, when performing effect processing on previously prepared acoustic signal, the pitch P of the input signal at a future time _I can also be used for processing. In this case, relates to the aforementioned [Delta] w (n), when P _I for the same n consecutively [Delta] w (n) or longer satisfies the above Equation 1, at all time range that continuous, corresponding to the input signal It is also conceivable to estimate that the pitch of the scale sound to be performed is n. In this case, in the example shown in FIG. 5, the pitch of the scale sound corresponding to the input signal is k from time t1 to time t3, and the scale sound corresponding to the input signal is from time t4 to time t6. Is assumed to be k−1.
Further, when P _I for the same n continuously from a certain time point [Delta] w (n) only future time satisfies the above Equation 1, at which point, the pitch of the chromatic note corresponding to the input signal is a n It is also conceivable to make an estimation.

  Next, the pitch processing unit 46 obtains a pitch shift amount with respect to the input signal, adds this to the pitch of the input signal acquired by the pitch detection unit 41, generates pitch information indicating the pitch of the voice signal, and performs pitch conversion. The unit 47 has a function used for pitch conversion processing. At this time, the pitch shift amount may be determined so that the pitch of the voice signal is closer to the pitch of the scale sound estimated by the scale sound estimation unit 43 to correspond to the input signal than the pitch of the input signal. Further, only when the scale sound estimation unit 43 can estimate the scale sound corresponding to the input signal, the pitch shift amount is obtained, and the pitch conversion unit 47 generates the voice signal. These pitch processing processes will be described in detail later.

  The pitch conversion unit 47 uses the pitch information of the input signal acquired by the pitch detection unit 41 and the pitch information of the voice signal generated by the pitch processing unit 46 to perform a pitch conversion process on the input signal to obtain a voice signal. Is a pitch conversion means having a function of generating. At this time, it is preferable to perform processing that converts only the pitch without changing the timbre as much as possible. The pitch conversion process employed in this embodiment will be described in detail later. In addition, the pitch conversion unit 47 supplies the generated voice signal to the mixing unit 60.

  Returning to the description of FIG. 2, the delay processing unit 50 is configured by a buffer memory or the like, and delays the input signal input to the mixing unit 60 by a time necessary for the voice signal generation processing in the voice signal generation unit 40. Means. The length of this delay may be about 20 milliseconds (ms), for example.

The mixing unit 60 is a mixing unit that mixes and outputs an input signal and a voice signal, and includes gain adjusting units 61 and 64, pan adjusting units 62 and 65, and adding units 63 and 66. The gain adjustment units 61 and 64 perform gain adjustment on the input signal delayed by the delay processing unit 50 and the voice signal generated by the voice signal generation unit 40, respectively, and then the pan adjustment units 62 and 65. Are assigned to the L side signal and the R side signal, added by the adders 63 and 66, and output as an L and R stereo signal.
Note that gain adjustment and pan adjustment are not essential, and an input signal and a voice signal may be simply added and output.

Of course, the signal processing unit 19 may be provided with functions of other effectors and mixers, and the output of the doubling effector 30 is input to these effectors for further effects, or input to the mixer. It can also be used for mixing processing. Conversely, it is also conceivable to input a signal after processing by another effector or mixer to the doubling effector 30.
In addition, when the signal processing unit 19 performs signal processing on a plurality of channels, it is needless to say that an effector may be provided for each channel and operated independently.

Next, the pitch detection process in the pitch detection part 41 is demonstrated using FIG.
In the pitch detection unit 41, basically, the pitch is detected by multiplying the input signal waveform 101 and the + side and − side envelopes of the input signal waveform 101 by a predetermined value (or a predetermined function value). This is done by detecting the timing at which the obtained + side envelope 102 and − side envelope 103 intersect (the magnitude relationship between the sample values is switched).

More specifically, a detection flag IRQ is prepared, and when the input signal waveform 101 intersects with the + side envelope 102, this is raised from 0 to 1 at the time points T ₁ and T ₃ , and the input signal waveform 101 becomes the − side envelope 103. At time points T ₂ and T ₄ when crossing, the time is decreased from 1 to 0, and the time from the rising edge of the IRQ flag to the next rising edge is measured by counting the number of samples. The measured time (number of samples) is taken as the detected pitch.

Although intersects the input signal waveform 101 is the + side envelope 102 immediately after T _1, since at this point the IRQ is already 1, the rise does not occur. Then, after the IRQ flag falls at T ₂ , the next rise occurs at T ₃ where the input signal waveform 101 intersects the + side envelope 102. The same is true for the negative envelope 103.
The use of the envelope in this way is the pitch for signals that contain many harmonic components and repeat zero crosses many times within one period, or signals that have a number of peaks due to their waveform being deformed. In order to prevent detection errors, a much more accurate pitch can be obtained compared to detection of only zero cross.

Here, the sampling frequency of the audio signal to be processed is 44.1 kilohertz (kHz), and in this case, the sampling period is about 0.02 ms. When it is desired to obtain the pitch with a finer accuracy than the sampling period, it is conceivable to perform interpolation to obtain the intersection timing more finely.
In the example shown in FIG. 5, the + and − envelopes 102 and 103 are attenuated as time elapses, the former is attenuated by the rising edge of the IRQ flag, and the latter is triggered by the falling edge. Is like resetting.

  Then, the pitch detection unit 41 sequentially records the pitch of the input signal detected by the operation as described above in the pitch buffer 42, and at a predetermined timing, here, every 6 ms, the pitch recorded in the pitch buffer 42 is recorded. Among them, a predetermined number of average values are output as pitch information indicating the pitch of the input signal waveform 101 at that time. Further, the predetermined number may be set to 16, for example, and if the number recorded in the pitch buffer 42 is less than this, the average value for the amount already recorded may be used.

  In addition, when performing the above detection, in order to remove noise and improve accuracy, and to distinguish a portion that should generate a voice signal from a portion that should not be generated, detection conditions and detection results are as follows. It is recommended to make a proper evaluation.

First, the pitch detection may be performed only when the level of the input signal is equal to or higher than a predetermined value. This is because a signal whose level is too low is considered as noise mixed in a silent signal.
Also, the number of zero crossings where the input signal waveform 101 crosses the zero level is counted, and when the number of zero crosses per time is equal to or greater than a predetermined value, it is preferable not to perform pitch detection. Here, the threshold value is 30 times or more per 6 ms. In such a part, the input signal corresponds to a consonant of human voice, and it is empirically found that it is preferable to add the voice signal in such a part without adding the voice signal. Therefore, the pitch detection is stopped and the generation of the voice signal is stopped in conjunction with the detection.

  Even when the pitch detection is started for an input signal satisfying the above criteria, the continuous detection mode is not used until the continuously detected pitch variation is within a predetermined range, for example, 12.5%. It is preferable that the detected pitch value is not recorded in the pitch buffer 42 until this condition is satisfied. This is because the detection result cannot be trusted when the error is large.

Further, the length of the period from the rise to the fall of the IRQ flag is measured as PCNT1, the length of the period from the fall to the rise is measured as PCNT0, and the following values (a) to (c) are obtained and buffered: When the error when the latest detected value is compared with the value previously recorded in the pitch buffer 42 is within a predetermined range, for example, within 12.5%, the detected value is newly set. You may make it record on the pitch buffer 42. FIG. Recording may be performed when the error is within a predetermined range for any number of (a) to (c).
(A) PCNT1 + PCNT0 (from the rise of the IRQ flag to the next rise)
(B) PCNT0 + PCNT1 (from the fall of the IRQ flag to the next fall)
(C) PCNT1 + PCNT0 for two cycles
(A) is the detected pitch value itself shown in FIG.

Also, instead of or in addition to the above (a) to (c), PCNT1 + PCNT0 for two cycles is detected as (d) and compared with twice the value of (a), and a cycle error is confirmed. You may do it.
Further, when the error is not within the predetermined range in the above (a) to (d), the number of times is counted as a detection error, and when this exceeds the predetermined number, the detection is stopped and the process is started again from the beginning. It may be.
For example, if the miss is 3 times or less, the pitch detection is continued without simply recording in the pitch buffer 42, and if the miss is 4 to 7 times, the detected value is recorded in the pitch buffer 42, The value to be compared is updated and the pitch detection is continued. If there are more than eight mistakes, all the data recorded in the pitch buffer 42 is discarded and the detection is restarted from the beginning.

Next, the pitch conversion process in the pitch conversion unit 47 will be described with reference to FIGS. 6 and 7.
In the pitch conversion unit 47, as the pitch conversion process, the input signal 111 is cut out using a window function, arranged as an element, and a process of determining the converted waveform pitch according to the arrangement period is performed. Here, as shown in FIGS. 6 and 7, the input signal 111 is cut out by shifting the timing of the two systems OUT0 and OUT1, and the sum of these signals is output as a voice signal after pitch conversion. I have to. According to such processing, pitch conversion can be performed while maintaining the formant information of the input signal 111.
This method is called the Lent method and is an application of the method described in the following paper.
Keith Lent (1989) “An efficient method for pitch shifting digitally sampled sounds.” Computer Music Journal Vol. 13 No.4. Pp.65-71

FIG. 6 shows a processing example in the case of pitch down (frequency decrease), and FIG. 7 shows a processing example in the case of pitch up (frequency increase).
Further, in these figures, P _I, the pitch value of the input signal 111 pitch detector 41 is output as the detection result, the P _V, a value of the pitch of the voice signal pitch processing unit 46 outputs. SB and RB indicate the lengths of the reference section and the output section, respectively. These codes are also used as codes representing the sections themselves. Further, since each of the above values changes depending on the content of the signal, “′” and “″” are added to the values at different time points to distinguish them.

Then, the pitch conversion processing, first, regardless of the output of the voice signal, writes the input signal 111 to the buffer, and set the reference section SB having twice the period of the pitch P _I in order for the input signal 111 I am going to go. Then, when performing extraction using a window function for output, this reference interval is used as a unit.
The length of the reference interval, of course vary but if Kaware the pitch P _I, because the pitch detector 41 as described above is to perform the output of the pitch information for each 6 ms, until the next output is performed, the value of the pitch P _I will not change.
Note that the above buffer may be shared with the buffer of the delay processing unit 50.

On the other hand, the generation of the output signal, first it starts the signal generation of OUT0 system, in this case, to set the output interval RB having twice the period of the pitch P _V. Then, in the output section RB, the input signal 112 in the latest reference section SB at the start time of the output section RB is read from the buffer sequentially and output. At this time, the read signal is multiplied by the window function 113. Here, a Hanning window having a length equal to the reference interval SB for reading is used as the window function. Further, the writing of the input signal 111 to the buffer and the reading for output are performed in parallel.

In the case of pitch down, since RB> SB, the output section RB continues even after all of the input signals 112 of the corresponding reference section SB are read out. Is output.
When the output section RB ends, a new output section RB ″ is set according to the pitch P _V ″ of the voice signal at that time, the input signal of the latest reference section SB ′ at the start time is read, and thereafter This process is repeated.

For even OUT1 system of signal generation, except for shifting the start time only P _V are the same as the processing in the case of OUT0 system. However, since the reference section for reading and the length of the output section are determined according to the information at the time of setting each output section, the same signal as the output signal of the OUT0 system is not always generated.
Then, as described above, the outputs of the OUT0 system and the OUT1 system are added and output as a voice signal. By such processing, a voice signal having a pitch _Pv having the same formant as the input signal 111 can be output.

  On the other hand, in the case of pitch up, since RB <SB as shown in FIG. 7, the output section RB ends before all the input signals 112 in the reference section SB are read. In this case, in the output section RB Reading is stopped at the end. The waveform indicated by the phantom line in FIG. 7 is a portion that is not read out thereafter. Correspondingly, a Hanning window having a width equal to RB is used as the window function 114.

However, at the end of the output section RB, the next output section RB ″ is set and reading of the input signal of the latest reference section SB at the start time is the same as in the case of FIG. As shown in the example of FIG. 7, when the reference section SB is not completed in the input signal 111 when the output section RB ″ is set, the input signal 112 of the same reference section SB is read again in the output section RB ″. .
Similarly to the case of FIG. 6, the start time is shifted by _Pv for OUT1 system signal generation.

By such processing, a voice signal having a pitch _Pv having the same formant as that of the input signal 111 can be output even in the case of pitch up.
Note that either processing can be applied if the pitch of the voice signal is made equal to the input signal.

In the pitch conversion process described above, the input signal 111 may be written to and read from the buffer at the same speed (number of processed samples per hour). It is also conceivable to obtain a gender effect that is converted from female to female or vice versa.
Furthermore, when a signal value at the timing between samples is necessary, such as when the pitch is not an integral multiple of the number of samples, it is preferable to perform interpolation processing as appropriate.

  Next, the processes related to pitch detection, scale sound estimation, pitch processing, and pitch conversion, which have been described above, executed by the CPU 11 will be described with reference to FIGS. Here, it is assumed that the CPU 11 acquires necessary information from the signal processing unit 19 and performs the processing. In this case, when the setting for enabling the function of the doubling effector 30 is made, the CPU 11 starts the processes shown in the flowcharts of FIGS. 8 to 10, 15 and 16 independently. However, these processes may be performed on the signal processing unit 19 side.

First, FIG. 8 shows a flowchart of the pitch detection process.
In this process, first, an input signal to be subjected to a doubling process is recorded in the input signal buffer and the output signal buffer for one sample (S11). Here, the input signal buffer is a buffer used for generating a voice signal in the pitch converter 47, and may be a ring buffer having a capacity for storing data of about 100 ms. The output signal buffer is a buffer used for delay processing by the delay processing unit 50, and may be a ring buffer having a capacity for storing data for about one second.

Thereafter, the zero cross of the input signal is counted (S12) and the sample counter is counted up (S13).
After that, if the pitch is being detected (S14), it is determined whether or not the input signal is the start position of the cycle (S15). Whether or not the pitch is being detected can be determined based on the contents of the pitch flag set in the process shown in FIG. Further, as described with reference to FIG. 5, the determination in step S15 can be performed by changing the IRQ flag according to the detection of the intersection between the input signal and the envelope and detecting the presence or absence of the rise.

If the result of the determination is the start position of the cycle (S16), the current value of the sample counter is recorded as pitch data in the pitch buffer 42 (S17), and the sample counter is reset (S18). Then, it waits until the next sample timing of the input signal (S19), returns to step S11 at the next sample timing, and repeats the processing.
On the other hand, if it is not the start position of the cycle in step S16, the process proceeds to step S19 and waits until the next sample timing. If the pitch is not being detected in step S14, the sample counter is reset (S18) and waits until the next sample timing (S19).

  In the process shown in FIG. 8, as described in the description of FIG. 5, it is preferable to perform various studies on the detection condition and the detection result when recording the pitch data in step S17. In order to simplify the explanation, the processing related to such examination is not shown here. Whether or not to detect the pitch is determined based on the number of zero crosses of the input signal by the processing shown in FIG.

Next, FIG. 9 shows a flowchart of the pitch detection control process.
In this process, first, when the number of zero crosses counted in step S12 in FIG. 8 is equal to or less than a predetermined value (here, 30 times as described above) (S21), the pitch flag is set to “1”. The pitch detection execution is shown (S22). Thereafter, the number of zero crosses is reset (S25), and the process waits until a predetermined time (here, 6 ms) elapses from the process of step S21 (S26), and then returns to step S21 to repeat the process.
If the number of zero crosses is greater than or equal to the predetermined value in step S21, the pitch flag is set to "0", indicating that the pitch detection is stopped (S23), and the pitch data recorded in the pitch buffer 42 is cleared. (S24), the process proceeds to step S25 and subsequent steps.

Therefore, in the process of the flowchart of FIG. 9, the process of step S21 is performed every predetermined time, and the pitch detection execution is set when the number of zero crosses during that time is equal to or smaller than the predetermined value, and when it is larger than the predetermined value, the pitch detection is stopped. Will be set.
With the processing shown in FIGS. 8 and 9, the pitch detection unit 41 can detect and control the pitch. However, the pitch value finally output as a detection result from the pitch detection unit 41 is a value obtained by the processing of the next FIG.

Next, FIG. 10 shows a flowchart of scale sound estimation and pitch setting processing.
In this process, first, the value of the pitch P _I of a predetermined number (e.g. 16) input signal the average value of the data of the pitch data recorded in the pitch buffer 42 (S31). Here, this value is the value of the pitch of the input signal output as a detection result from the pitch detection unit 41, and this value is also used in the pitch conversion processing in the pitch conversion unit 47.

Then, based on this value of the pitch P _I, to obtain a pitch closest chromatic note to the pitch P _I in which stores to the pitch table 44 (S32). Then, it is determined that the difference between the pitch of the pitch P _I and their chromatic note unless the predetermined error Δh (n) / 2 less (S33), scale notes corresponding to the current input signal is not an estimate buffer At the same time, the timer for measuring the duration of the pitch is stopped (S35). Then, clear the pitch P _V of the voice signal (S39), as described in the description of a pitch conversion processing later Thus, it stops the output of the voice signal. And it waits until predetermined time (here 6 ms as mentioned above) passes since the process of step S31 (S44), returns to step S31, and repeats a process after that. In the case where appropriate P _I was not determined in the step S31 does not get a pitch in step S32, may determine in step S33 is made to be to NO.

On the other hand, if “YES” in the step S33, it can be determined that there is a scale sound corresponding to the current input signal. The pitch of the obtained chromatic notes in step S32 is, if other than the estimated chromatic note pitch P _E stored in the estimated value buffer 45 (S36), registers the pitch acquired in step S32 to estimate the buffer 45 In step S37, the timer is reset to start measuring time (step S38), and the process proceeds to step S39 and subsequent steps.
If YES in step S36, if the measured time of the timer has elapsed by a predetermined time Δw (n) (S40), the pitch P _{I of the} input signal continues for the predetermined time and the estimated scale pitch P Since it can be seen from _E that it was within the predetermined error range, it is estimated that the scale sound corresponding to the input signal at that time is the scale sound whose pitch was acquired in step S32.

Then, the pitch is registered in the estimated value buffer 45, the estimated chromatic note pitch P _E is the pitch of the sound scale that is estimated to correspond to the input signal with (S41), to perform the generation of the voice signal, input A pitch shift amount ΔP is determined from the signal pitch P _I and the estimated scale pitch P _E (S42), and a value obtained by adding this to the pitch P _{I of the} input signal is defined as the pitch P _{V of the} voice signal (S43). The P _V is used in a pitch conversion processing in the pitch conversion unit 47.

Then, it waits until predetermined time passes from the process of step S31 (S44), returns to step S31, and repeats a process.
If NO in step S40, it is not possible to distinguish whether the input signal corresponds to a scale sound or if it just happens to have a pitch close to a scale sound, so the process proceeds to step S39 to output a voice signal. And the subsequent processing is continued.

Through the processing shown in FIG. 10, the pitch of the input signal is obtained, the scale sound corresponding to the input signal is estimated based on the pitch, and the pitch conversion unit 47 uses the estimated pitch of the scale sound. It is possible to generate and set the pitch of the voice signal used for the pitch conversion process in FIG. These pitches are updated every predetermined time.
Of the processes shown in FIG. 10, the process of step S31 is a process corresponding to the function of the pitch detection unit 41, steps S32 to S41 are the processes corresponding to the function of the scale sound estimation unit 43, and steps S42 and S43 are the pitch. This is processing corresponding to the function of the processing unit 46.

Next, FIG. 11 shows a flowchart of a first example of the pitch shift amount determination processing shown in step S42 of FIG.
In this process, first, | P _E −P _I | × 2 is obtained, and the smaller one of this value and the positive constant C is set to ΔP ″ (S51). A value obtained by multiplying by a random number R in the range of 1 and passing through a low-pass filter (LPF) is set as ΔP ′ (S52), where LPF has a cutoff frequency of 0.1 Hz to 10 Hz and a gain of 1 at maximum. Should be used.

Thereafter, if P _E > P _I , ΔP = ΔP ′, otherwise ΔP = −ΔP ′ and ΔP is obtained (S53 to S55), and the process returns to the original process.
Because it is [Delta] P '≧ 0, the pitch shift amount [Delta] P obtained by these processes, a shift amount as close pitch P _I on the estimated chromatic note pitch P _E of the input signal.

12, when determining the ΔP in the process shown in FIG. 11 shows an example of the relationship between _{P I} and _{P V.}
This figure shows an example in which the pitch of the input signal changes as indicated by a solid line 81, and the input signal is estimated to correspond to the scale sound having the pitch indicated by the thick line 82 in the period indicated by the thick line 82. ing. In this example, C> | P _E −P _I | × 2 in the sections indicated by the symbols a and c, and C <| P _E −P _I | × 2 in the section indicated by the symbols b.

In such a case, P _V = P _I + ΔP at each time point is located somewhere in the range indicated by the arrows between the solid line 81 and the broken line 83 according to the value of the random number R and the processing result of the LPF. Will do. Therefore, by determining the ΔP by the process of FIG. 11, the _{P V,} always _| P I -P _{E |} ≧ _| it can be defined such that | P V -P _E. That is, it can be said that the pitch of the voice signal generated by the pitch processing unit 46 can be made closer to the pitch of the scale tone than the pitch of the corresponding input signal.

FIG. 13 shows a flowchart of a second example of the pitch shift amount determination process shown in step S42 of FIG.
In this processing, for a positive constant C and a random number R in the range from 0 to 1, C × (R−0.5) is filtered with an LPF or a bandpass filter (BFP), and ΔP ′ (S61), ΔP is obtained as ΔP = P _E −P _I + ΔP ′ (S62), and the process returns to the original process. As the LPF or BPF, for example, a cutoff frequency or center frequency of 0.1 to 10 Hz and a maximum gain of 1 may be used.

14, when determining the ΔP in the process shown in FIG. 13 shows an example of the relationship between _{P I} and _{P V.}
The input signal and scale sound shown in this figure are the same as those shown in FIG. However, in this example, P _V = P _I + ΔP at each time point indicates a constant C width arrow between the broken line 84 and the broken line 85 according to the value of the random number R and the result of the filtering process. It will be located somewhere in the range.

As can be seen from this figure, when P _I is close to P _E , P _V may partially be farther from P _E than P _I , but ΔP is determined by the process of FIG. However, as a whole, it can be said that the pitch of the voice signal generated by the pitch processing unit 46 can be made closer to the pitch of the scale tone than the pitch of the corresponding input signal.

Next, FIG. 15 shows a flowchart of the reference section setting process.
In this process, first, it is determined whether or not either a reference section is set at the reference position or the reference position has reached the end of the reference section (S71).
If not satisfied, the reference position of the input signal recorded in the input signal buffer is advanced by one sample (S74), waits for the next sample timing (S75), and then returns to step S71. Repeat the process.

On the other hand, if either has been met at step S71, if the pitch P _I of an input signal by the processing of step S31 in FIG. 10 is set at that time (S72), the input stored in the input signal buffer the signal, and the start position of the current reference position, and sets the next reference period to the length SB and 2 times the pitch P _I (S73), the process proceeds to step S74, the continued following process. If it is not set in step S72, the process proceeds to step S74 as it is and the following processing is continued.
With the processing shown in FIG. 15 described above, the reference interval as described with reference to FIGS. 6 and 7 can be set for the input signal to be processed in the pitch conversion unit 47. The “reference position” is simply used to detect the end of the reference section, and any parameter can be used as long as it can measure the progress of the process.

Next, FIG. 16 shows a flowchart of the pitch conversion process.
In this process, first, it is determined whether or not any output section is set or whether the reading position is advanced in the current output section to the end of the output section (S81).
If not satisfied, it is determined whether or not the reading position has exceeded the end of the reference section (S85). If not, one sample of data at the reading position is read from the input signal buffer. The value of the window function corresponding to the reading position is multiplied and output as voice signal data (S86). If it exceeds, 0 is output (S87). In either case, the reading position is advanced by one sample (S88). Note that the window function is the same as described with reference to FIGS.

If the pitch is being detected (pitch flag is “1”) (S89), the process waits until the next sample timing (S90), and then returns to step S81 to repeat the process. On the other hand, if being detected pitch, clear the output section is configured (S91), then waits until the pitch P _V of the voice signal is set (S92), the pitch P _V is set Returning to step S81, the process is repeated. That is, the output of the voice signal is stopped until the pitch _Pv is set again.

Also, if YES in step S81, it is determined whether or not the pitch P _V of the voice signal in the process of step S42 in FIG. 10 is set (S82), if set, the next output period length RB and set to 2 times the pitch _{P V} with (S83), the read position, and moved to the start position of the latest reference section of the processing time (S84), the process proceeds to step S85 following process. On the other hand, if not set in step S82, the process proceeds to step S91 and subsequent steps. That is, also in this case, the output of the voice signal is stopped until the pitch _Pv is set again.

With the processing shown in FIG. 16 described above, the pitch converter 47 can generate an output signal of the OUT0 system as described with reference to FIGS. 6 and 7 based on the input signal to be processed. Then, as described above, a voice signal can be generated by adding this output signal and the OUT1 system output signal. Generation processing of the output signal of the OUT1 line, the start timing except shifted by the pitch P _V a, but those same as the processing shown in above FIG. 16, the pitch P _V settings lost or pitch detection of There so that shifting the timing initiated when resume output after or aborted, after step S92, it is preferable to add a standby processing of the pitch P _V min.

In the electronic musical instrument 10, when the CPU 11 and the signal processing unit 19 execute the above-described processing, the scale sound corresponding to the input signal is estimated and the scale sound corresponding to the input signal can be estimated. The voice signal can be generated by performing pitch conversion processing so that the pitch of the input signal approaches the pitch of the scale sound, and the input signal and the voice signal can be mixed and output.
Therefore, it is possible to prevent the pitch of the voice signal from greatly deviating from the pitch of the scale sound, and to prevent the output sound from becoming unnaturally audible due to the superposition thereof.

Further, when the difference between the pitch of the input signal and the pitch of a certain scale sound is within a predetermined error range, the scale sound is estimated to be a scale sound corresponding to the input signal, and the input signal Since the voice signal is generated only when the corresponding scale sound can be estimated, it is possible to emphasize the input signal only in the portion close to the pitch of the scale sound and make the output sound natural.
These features can be combined to obtain a suitable effect synergistically, but each also exerts an effect.

That is, for example, in step S42 in FIG. 10, it is not particularly conscious that the pitch P _V of the voice signal is close to the estimated scale pitch P _E. A value obtained by filtering (R−0.5) with an LPF or a bandpass filter (BFP) may be ΔP. Also in this case, the LPF and BPF may be the same as those used in the processing of FIG.

Further, step S42 without clearing the pitch _{P V} of the voice signal at step S39 of FIG. 10, a value obtained by filtering with C × (R-0.5) the LPF or bandpass filters (BFP) as ΔP You may make it progress to. That is, even when the scale sound corresponding to the input signal cannot be estimated, a signal obtained by randomly changing the pitch of the input signal by the width of the constant C may be superimposed as a voice signal.
In this case, if ΔP is prominently discontinuous border with the ΔP interval seeking at intervals and step S39 to determine a ΔP at the step S42, in order to mitigate a discontinuity, performing cross-fade for ΔP or P _V treatment May be added.
Incidentally, the method of determining the pitch P _V of the voice signal is not limited to these, of course.

[Second Embodiment: FIGS. 17 to 20]
Next, an electronic musical instrument which is a second embodiment of the acoustic signal processing apparatus of the present invention will be described. However, since this electronic musical instrument is the same as the electronic musical instrument of the first embodiment except that the configuration of the voice signal generation unit is slightly different, the description other than this point is omitted. Moreover, the same code | symbol is used about the structure corresponding to 1st Embodiment. Further, in this embodiment, when effect processing is performed on an input signal prepared as data in advance, that is, an input signal that can also refer to the contents of a time point before the time point being processed, instead of an input signal input in real time. This is a preferred embodiment.

First, FIG. 17 shows a configuration of a voice signal generation unit in the electronic musical instrument of this embodiment.
As shown in this figure, the voice signal generator 40 'in this embodiment has a phrase detector 48 in addition to the same configuration as in the first embodiment. Further, the content of the pitch machining process in the pitch machining unit 46 ′ is also different from that in the first embodiment.
And the phrase detection part 48 is a detection means which has a function which detects the position of the phrase in an input signal. This phrase refers to a series of pronunciation intervals that are almost uninterrupted. For example, in the case of a human voice, this phrase corresponds to a portion uttered at a breath. Then, the phrase detection unit 48 transmits the phrase detection result to the pitch processing unit 46. The pitch processing unit 46 obtains the pitch shift amount for the input signal in units of phrases based on the information from the phrase detection unit 48, and adds this to the pitch of the input signal acquired by the pitch detection unit 41 to obtain the voice signal. Pitch information indicating the pitch of is generated.

Next, processing executed by the CPU 11 in the electronic musical instrument of this embodiment will be described.
Also in this electronic musical instrument, the processing to be executed by the CPU 11 (or the signal processing unit 19) is substantially the same as that described with reference to FIGS. 8 to 16 in the first embodiment. However, the processing is performed on an input signal prepared as data in advance, and the processing timing is controlled by using the number of samples by converting the timing into the number of samples based on the sampling period of the input signal. ing. Therefore, in the process of waiting for a predetermined time, it is only necessary to wait until the processing of the number of samples corresponding to the time is completed, and an event that has occurred at a certain timing can be performed without performing each process in synchronization. If it is stored in correspondence with the position of the sample, it can be processed later with reference to the information.
Here, only processing different from the case of the first embodiment except for this point will be described using a flowchart.

First, this embodiment is different from the first embodiment in that the phrase break detection process is executed between steps S11 and S12 in the pitch detection process shown in FIG.
FIG. 18 shows a flowchart of the phrase break detection process.
After the process of step S11 in FIG. 8, the CPU 11 determines whether the sample value of the input signal is equal to or less than a predetermined value (S101). If it is equal to or less than the predetermined value, if the silent section counter is not being counted (S102), the count is started (S106), and the fact that the timing of the sample being processed is in the phrase is recorded (S109). ), The process proceeds to step S12 in FIG. The silent section counter is a counter for counting the length of time that the volume level of the input signal continues below a predetermined value.

On the other hand, if the silent section counter is counting in step S102, the silent section counter is counted up (S103). If the count value is equal to or greater than a predetermined threshold (S104), the fact that the timing of the sample being processed is a phrase break is recorded (S105), and the process returns to the original process. If it is not greater than or equal to the threshold value in step S104, the fact that the corresponding timing is in the phrase is recorded (S109), and the process proceeds to step S12 in FIG.
If NO in step S101, the counter is reset if the silent section counter is counting, and the count is stopped, and if it is not counting, the fact that the corresponding timing is in the phrase is recorded. (S107 to S109), the process proceeds to step S12 in FIG.

This process is a process corresponding to the function of the phrase detection unit 48. When a period of a sample value of the input signal that is equal to or smaller than a predetermined value continues for a predetermined time or longer as a result of this process, this is regarded as a break between phrases of the input signal. Can be detected.
Note that the information about the break of the phrase and the portion in the phrase recorded in step S105 or S109 may be transmitted to the pitch processing unit 46 'for each sample timing one by one. The information may be transmitted as information on the start timing and end timing of each phrase.

In step S101, a volume envelope may be obtained instead of the sample value of the input signal, and it may be determined whether or not the volume indicated by the volume envelope is equal to or less than a predetermined value. In this case, in order to ensure the accuracy of detection, the volume envelope may be obtained after passing the input signal through some filter.
Further, the threshold used in step S104 may be a value indicating a time that can be recognized as a break of a phrase when a normal person hears it with an ear.

In this embodiment, scale sound estimation and pitch setting processing are also different from those in the first embodiment. That is, the pitch shift amount [Delta] P, because of the so determined for each phrase, the process of setting the pitch P _I of the scale sound estimation and the input signal, a processing for setting the pitch P _V of the pitch shift amount [Delta] P and the voice signal Are done separately.

FIG. 19 shows a flowchart of scale sound estimation and pitch setting processing in this embodiment. In this figure, the same step numbers are assigned to parts common to the processing in the first embodiment shown in FIG.
And differs from the process shown in the process 10, first, in step S31 ', the pitch P _I of an input signal, as the value of the processing time, as stored in correspondence with the timing information It is a point. The information of the timing, may be stored or to determine the P _I on the basis of the data in the state where the pitch detection process is completed for several samples of the input signal.
Further, the time measurement by the timer in step S38 or the like is not the actual time, but the reproduction time of the input data is measured based on the number of samples of the input signal, as in the case of timing management.

In the case of YES at step S40, as the estimated chromatic notes pitch P _E of a pitch processing timing registered in the estimated value buffer 45, in correspondence with the timing information so that they can be referenced later in the pitch processing and storing (SX), wait until the next processing timing (SY), and return to step S31 'to repeat the processing.
The timing information here is the same as the timing information used in step S31 ′, and the next processing timing is, for example, for the input signal for 6 ms as the reproduction time after the processing in step S31 ′. The timing at which the pitch detection process is completed can be set.

Further, in the case of NO in step S40 and after step S35 or S38, information indicating that the estimated scale pitch at the processing timing is “none” is stored in association with the timing information (SZ), and step Proceed to SY. The timing information here is also the same as the timing information used in step S31 ′.
By performing the processing as described above, the scale sound corresponding to the input signal at each timing can be estimated at appropriate intervals, as in the case of the processing shown in FIG. Then, the pitch of the scale sound can be stored so that it can be used in the pitch processing.

Next, FIG. 20 shows a flowchart of the pitch machining process in this embodiment.
In this embodiment, at the stage where the information of the pitch P _I and the estimated chromatic note pitch P _E of the input signal at each processing timing for one range of phrase content becomes available, the processing shown in FIG. 20, The pitch P _V of the voice signal at each processing timing within the range is obtained.

In this process, first, an average value of (P _E −P _I ) is obtained within a range where there is an effective estimated musical scale pitch P _E in the phrase to be processed, and this is set as ΔP ′ (S111). Then, for a positive constant C and a random number R in the range from 0 to 1, a pitch shift amount ΔP = ΔP ′ + C × (R−0.5) is set (S112). This ΔP may be common within the phrase, or may be obtained separately by re-taking random numbers at each processing timing. Then, using the [Delta] P, the pitch _{P V} of the corresponding voice signal _{P I} of the processing _timing, determined by _{P V = P I + ΔP (} S113), the process ends.
That is, P _V = P _I + [average value of (P _E −P _I )] + C × (R−0.5).

Here, the relationship between P _I and P _V when P _V is obtained by the process shown in FIG. 20 will be described with reference to FIG.
In this figure, the pitch of the input signal by a solid line 91, within the range of valid estimate chromatic note pitch P _E is present, shows this in bold lines 92 and 93. Then, the average value calculated by the step S111 in FIG. 20, of the period in the phrase, the period indicated by the thick line 92 and 93 only _adds the value of _(P E -P _I), the number of data of the period It can be obtained by dividing. In addition, an average value is separately obtained for the period indicated by the thick line 92 and the period indicated by the thick line 93, and these average values are also obtained by weighting according to the length of each period and further averaging. Can do.

In the case of C = 0, the P _V is simply becomes a value obtained by adding the average value to P _I, takes a value indicated by the dashed line 94 is moved parallel to the solid line 91. Further, when C is not 0, P _V is centered on the dashed line 94, takes a value in a region having a width C which is sandwiched between the dashed line 95, 96.
Then, as can be seen from this figure, even when so determined by the processing shown the P _V in FIG. 20, but not necessarily true for individual processing timing, the P _E than P _I as a whole it is possible to find the nearby made such a P _V. That is, it is possible to generate a voice signal having a pitch closer to the scale sound than the input signal. As a result, even with the electronic musical instrument of this embodiment, the pitch of the voice signal is prevented from greatly deviating from the pitch of the scale sound, and the superimposition of the output sound makes the sound unnatural. Can be prevented.

Also in this embodiment, as in the first embodiment, only when the P _E is a valid value, it may be performed to generate the voice signal. In this way, it is possible to enhance the input signal only in the part close to the pitch of the scale sound and make the output sound natural.
However, conversely, it is also conceivable that the voice signal is generated even at the break of the phrase by continuously using ΔP obtained in the phrase until the next phrase starts.
Alternatively, the phrase may be detected as a break between phrases when the state in which the pitch detection by the pitch detector 41 cannot be performed properly continues for a predetermined time or longer. In this case, the pitch detection unit 41 may input the pitch detection result of the input signal to the phrase detection unit 48.

Although the description of the embodiment has been completed above, it is a matter of course that the configuration of the apparatus, specific processing contents, and the like are not limited to those described in the above-described embodiments.
For example, when estimating the scale sound corresponding to the input signal, the two steps of searching for a scale sound with a close pitch and confirming whether the pitch difference is within a predetermined error are not performed, A pitch range is determined in consideration of errors, and when the pitch of the input sound belongs to any of them, it is assumed that the scale sound corresponding to the input sound is a scale sound having the pitch range. Good. Also, the scale sound does not necessarily have to belong to the scale of equal temperament.

In the above-described embodiment, the Lent method is adopted for the pitch conversion processing. However, pitch conversion may be performed by a method other than this. Furthermore, the processing target may be an analog acoustic signal, and pitch detection processing, pitch processing processing, pitch conversion processing, mix processing, and the like may be performed by an analog circuit.
Furthermore, these modifications and the contents described in the first and second embodiments can be appropriately combined within a range in which no contradiction occurs.

  Of course, the present invention can be applied to an acoustic signal processing apparatus other than an electronic musical instrument. Any device can be used as long as it has a function of handling the acoustic signal shown. Furthermore, the present invention can be implemented as a program for providing an effector function to a single effector or apparatus.

  Further, the program of the present invention is a program for causing a computer to control hardware so as to function as the above-described acoustic signal processing apparatus. In addition to being stored in advance in a ROM, HDD, etc., a CD-ROM or flexible The program is recorded on a non-volatile recording medium (memory) such as a disk, and this program is read from the memory to the RAM and executed by the CPU, or an external device or program including the recording medium on which the program is recorded is stored in the HDD or the like. The same effect can be obtained even when downloaded from an external device stored in the storage means and executed.

As is clear from the above description, according to the acoustic signal processing apparatus or program of the present invention, it is possible to realize an effect capable of always obtaining an output signal with a natural audibility.
Therefore, according to the present invention, it is possible to provide an acoustic signal processing device capable of generating natural and novel sounds.

1 is a block diagram showing a configuration of an electronic musical instrument which is a first embodiment of an acoustic signal processing device of the present invention. It is a figure which shows the function structure of the doubling effector with which the signal processing part shown in FIG. 1 is equipped. It is a figure which shows the function structure of the voice signal production | generation part shown in FIG. It is a figure which shows the example of the estimation result at the time of estimating the scale sound corresponding to an input signal in the scale sound estimation part shown in FIG. It is a figure for demonstrating the pitch detection process in the pitch detection part shown in FIG.

It is a figure for demonstrating the pitch conversion process in the pitch conversion part shown in FIG. It is another figure. It is a flowchart of the pitch detection process which CPU of the electronic musical instrument shown in FIG. 1 performs. It is a flowchart of a pitch detection control process similarly. It is a flowchart of a scale sound estimation and pitch setting process similarly.

It is a flowchart of the 1st example of the pitch shift amount determination process performed by step S42 of FIG. When determining the ΔP in the process shown in FIG. 11 is a diagram showing an example of the relationship between P _I and P _V. It is a flowchart of the 2nd example of pitch shift amount determination processing. When determining the ΔP in the process shown in FIG. 13 is a diagram showing an example of the relationship between P _I and P _V. It is a flowchart of the reference | standard area setting process which CPU of the electronic musical instrument shown in FIG. 1 performs. It is a flowchart of a pitch conversion process similarly. It is a figure which shows the function structure of the voice signal production | generation part in the electronic musical instrument which is 2nd Embodiment of the acoustic signal processing apparatus of this invention. It is a flowchart of the phrase detection process which CPU of an electronic musical instrument performs in 2nd Embodiment. It is a flowchart of a scale sound estimation and pitch setting process similarly. It is a flowchart of a pitch process similarly. The processing shown in FIG. 20 of the case of obtaining the P _V, which is a diagram for explaining a relationship between P _I and P _V.

Explanation of symbols

DESCRIPTION OF SYMBOLS 10 ... Electronic musical instrument, 11 ... CPU, 12 ... ROM, 13 ... RAM, 14 ... Detection circuit, 15 ... Display circuit, 16 ... Audio signal I / F, 17 ... Communication I / F, 18 ... Sound source part, 19 ... Signal Processing unit 20 ... System bus 21 ... Operator 22 ... Indicator 23 ... Sound system 30 ... Doubling effector 40, 40 '... Voice signal generation unit 41 ... Pitch detection unit 42 ... Pitch buffer 43 ... scale sound estimation unit, 44 ... pitch table, 45 ... estimated value buffer, 46, 46 '... pitch processing unit, 47 ... pitch conversion unit, 48 ... phrase detection unit, 50 ... delay processing unit, 60 ... mix unit, 61, 64 ... gain adjustment unit, 62, 65 ... pan adjustment unit, 63, 66 ... addition unit

Claims (2)

Processing signal generating means for generating a processing signal by performing pitch conversion processing on the input signal;
Mixing means for mixing and outputting the input signal and the processing signal;
Pitch storage means for storing the pitches of all scale sounds within a predetermined range of a certain scale;
Estimating means for estimating a scale sound corresponding to the input signal,
The processing signal generating means only when said estimating means could estimate the chromatic note corresponding to the input signal, row the pitch conversion processing so as to be close to the pitch of the estimated chromatic note pitch of the input signal Ri means der for generating the machining signals,
When the estimation means has a difference between a pitch of the input signal and a pitch of a scale sound stored in the pitch storage means within a predetermined error range, the scale sound corresponds to the scale sound corresponding to the input signal. Is a means of presuming that
The predetermined error range is defined so that an error range for a certain scale sound does not overlap an error range for another scale sound for all the scale sounds for which the pitch storage means stores the pitch. An acoustic signal processing device. Computer
Processing signal generating means for generating a processing signal by performing pitch conversion processing on the input signal;
Mixing means for mixing and outputting the input signal and the processing signal;
Pitch storage means for storing the pitches of all scale sounds within a predetermined range of a certain scale;
A program for functioning as an estimation means for estimating a scale sound corresponding to the input signal,
The processing signal generating means only when said estimating means could estimate the chromatic note corresponding to the input signal, row the pitch conversion processing so as to be close to the pitch of the estimated chromatic note pitch of the input signal Ri means der for generating the machining signals,
When the estimation means has a difference between a pitch of the input signal and a pitch of a scale sound stored in the pitch storage means within a predetermined error range, the scale sound corresponds to the scale sound corresponding to the input signal. Is a means of presuming that
The predetermined error range is defined so that an error range for a certain scale sound does not overlap an error range for another scale sound for all the scale sounds for which the pitch storage means stores the pitch. Program.

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