JP2014071403A - Sound signal processing apparatus, and sound signal processing method - Google Patents

Abstract

PROBLEM TO BE SOLVED: To allow satisfactory interpolation of a sound signal.SOLUTION: A distorted waveform generation unit 2 generates a distorted waveform by outputting a signal following up a waveform of an original signal supplied to an input terminal 1 in a period when the waveform of the original signal is inclined upward, and by outputting a signal not including a prescribed high frequency of frequencies of the original signal in a period when the waveform of the original signal is inclined downward. A harmonics extraction unit 3 extracts a signal in a prescribed frequency band from the distorted waveform generated by the distorted waveform generation unit 2, as harmonics of the original signal. An adder 5 adds the harmonics extracted by the harmonic extraction unit to the original sound to be supplied to the input terminal 1.

Description

  The present invention relates to an acoustic signal processing device and an acoustic signal processing method suitable for use in, for example, a digital audio device or a mobile phone terminal device.

  2. Description of the Related Art In recent years, a speaker has been incorporated in a housing of a docking station, which is a function expansion unit for a notebook personal computer, a multifunctional mobile phone terminal called a smartphone, a mobile phone terminal, or the like. A speaker built in such a small casing naturally becomes small in size. However, if the speaker is downsized, it may be difficult to obtain a sufficient low-frequency sound. In addition, it is known that the low frequency range is lost by performing compression using a compression technique such as MP3 (MPEG Audio Layer-3). It becomes difficult to obtain.

  For example, in Patent Document 1, a harmonic component is generated from a bass component below the lowest frequency that can be reproduced by a speaker, and the harmonic component is added to the original sound so that the bass component can be heard by the harmonic component. The technology to be described is described. This technology uses a phenomenon called “Missing fundamental”. “Missing fundamental” is a phenomenon that even if a frequency range including a fundamental frequency is removed from a certain sound, it is recognized as the same pitch as the original sound. This phenomenon is known to occur because the brain perceives sound by using not only the fundamental frequency but also the ratio of harmonics.

  In recent years, it has become popular to use audio data by distributing it via a network such as the Internet or recording it on a portable audio reproducing device or portable terminal device. In this way, in audio data distributed over a network or recorded on a recording medium, in order to avoid an increase in the amount of data and an increase in occupied bandwidth due to an excessively wide band, generally music to be supplied, etc. Among them, components having a certain frequency or higher are removed. For example, in audio data in MP3 (MPEG1 audio layer 3) format, frequency components of about 16 kHz or higher are removed, and in audio data in ATRAC3 (Adaptive TRansform Acoustic Coding 3) format, frequency components of about 14 kHz or more are removed. The In such a signal from which the high frequency component has been completely removed, the sound quality is slightly changed, and the sound quality is deteriorated as compared with the original music.

  For example, Patent Document 2 describes a technique for performing high-frequency signal interpolation by extracting a high-frequency component of a signal from a white noise generator and adding it to an original signal.

JP 2005-318598 A Japanese Patent Laid-Open No. 2-311006

  In the low-frequency interpolation technique described in Patent Document 1, harmonics of low-frequency components are generated by peak-holding and rectifying low-frequency components below the lowest frequency that can be reproduced by the speaker, and the generated harmonics are added to the original signal. Interpolation of low frequency signals. However, in the technique described in Patent Document 1, the held peak value is reset at the timing when the input signal transitions from positive to negative or from negative to positive. Since the harmonics are greatly generated at the moment when the held peak value is reset, the waveform of this portion is greatly deviated from the waveform of the original signal. When harmonics having a waveform different from the waveform of the original signal are added to the original signal, the sound quality of the reproduced acoustic signal is deteriorated.

  Further, in the high frequency interpolation technique described in Patent Document 2, since the high frequency component is extracted from the signal generated by the white noise generator, the extracted high frequency component is a signal having no correlation with the original signal. End up. Therefore, the sound quality of the acoustic signal after interpolation is degraded as compared with the sound quality of the original signal.

  The present invention has been made in view of the above situation, and an object of the present invention is to perform satisfactory acoustic signal interpolation.

  In order to solve the above problems, the acoustic signal processing device of the present invention is configured to include a distortion waveform generation unit, a harmonic extraction unit, and an adder. The distortion waveform generation unit outputs a signal that follows the waveform of the original signal during a period in which the slope of the waveform of the original signal supplied to the input terminal is upward, and a period in which the slope of the waveform of the original signal is downward A distortion waveform is generated by outputting a signal that does not include a predetermined high frequency among the original signals. The harmonic extraction unit extracts a signal in a predetermined frequency band as a harmonic of the original signal from the distortion waveform generated by the distortion waveform generation unit. The adder adds the harmonic extracted by the harmonic extraction unit to the original signal supplied to the input terminal.

  The acoustic signal processing method of the present invention first outputs a signal that follows the waveform of the original signal during a period in which the gradient of the waveform of the original signal supplied to the input terminal is upward. During the downward period, a distorted waveform is generated by outputting a signal that does not include a predetermined high frequency among the original signals. Subsequently, a signal in a predetermined frequency band is extracted from the distorted waveform as a harmonic of the original signal. Then, the extracted harmonic is added to the original signal supplied to the input terminal.

  By configuring the acoustic signal processing device as described above and performing acoustic signal processing, a signal in a predetermined frequency band is extracted as a harmonic of the original signal from a distorted waveform having a waveform corresponding to the slope of the waveform of the original signal. The That is, the extracted harmonics also have a waveform corresponding to the waveform of the original signal. Then, by adding such harmonics to the original signal, the sound quality of the interpolated acoustic signal becomes closer to the original signal and more natural and good.

  According to the acoustic signal processing device and the acoustic signal processing method of the present invention, it is possible to perform satisfactory acoustic signal interpolation.

It is a block diagram which shows the structural example of the acoustic signal processing apparatus by one embodiment of this invention. It is the graph which superimposed and showed the distortion waveform produced | generated by the distortion waveform production | generation part by the example of 1 embodiment of this invention on the waveform of an input signal. It is the graph which superimposed and showed the waveform of the signal which passed BPF by the example of 1 embodiment of this invention on the waveform of an input signal. It is a block diagram which shows the structural example of the distortion waveform generation part by one embodiment of this invention. FIG. 5A is a block diagram illustrating a configuration example of a distortion waveform generation unit according to an embodiment of the present invention, FIG. 5A illustrates a configuration example during operation as an LPF, and FIG. 5B illustrates a configuration example during operation as an APF. It is the graph which showed the envelope detected using Hilbert transform superimposed on the waveform of the input signal. It is the graph which superimposed and showed the distortion waveform produced | generated by the distortion waveform production | generation part by one embodiment of this invention on the waveform of an input signal. It is the graph which superimposed and showed the distortion waveform produced | generated by the distortion waveform production | generation part by one embodiment of this invention on the waveform of an input signal.

An example of a receiving device according to an embodiment of the present disclosure will be described in the following order with reference to the drawings. However, the present disclosure is not limited to the following example.
[1. Configuration example of acoustic signal processing apparatus]
First, an acoustic signal processing device according to an embodiment of the present disclosure will be described with reference to FIG. FIG. 1 is a block diagram showing the configuration of the acoustic signal processing apparatus 10. As shown in FIG. 1, an acoustic signal processing apparatus 10 according to the present embodiment includes an input terminal 1, a distortion waveform generation unit 2, and a bandpass filter (hereinafter referred to as “BPF”) as a harmonic extraction unit. ) 3, a delay circuit 4, an adder 5, and an output terminal 6.

  In FIG. 1, the input terminal 1 is supplied with a digital acoustic signal reproduced from a device with compression processing such as MP3 or ATRAC3 as an original signal. The original signal supplied to the input terminal 1 is supplied to the distortion waveform generator 2 and the delay circuit 4. The distortion waveform generator 2 operates as an all-pass filter (hereinafter referred to as “APF”) when the slope of the waveform of the input original signal is upward, and as a low-pass filter (hereinafter referred to as “LPF”) when the waveform is downward. By operating, a distorted waveform including an envelope waveform is generated. The distortion waveform generation unit 2 can be configured by, for example, a DSP (Digital Signal Processor), and can generate a distortion waveform of an acoustic signal on software.

  The BPF 3 extracts a signal in a predetermined frequency band from the distorted waveform including the envelope waveform input from the distorted waveform generation unit 2 and supplies the extracted signal to the adder 5. The pass band of BPF 3 can be set to a value corresponding to the target frequency to be interpolated. For example, when interpolating a low frequency, only the harmonics of the low frequency signal to be interpolated can be extracted by the BPF 3 by setting a band from the lowest reproduction frequency of the speaker to 2 or 3 times. It becomes possible. When interpolating high-frequency frequencies, it is possible to extract only the harmonics of the high-frequency signal to be interpolated by setting the band from the frequency cut by compression processing etc. to the front of the Nyquist frequency It becomes. In the case of interpolating a high frequency, the BPF 3 may be an HPF (High Pass Filter). In this case, it is possible to extract only harmonics in a frequency range higher than the frequency cut by the compression process or the like.

  FIG. 2 shows an example of a distortion waveform generated by the distortion waveform generation unit 2. The waveform indicated by the solid line in FIG. 2 is the waveform (input waveform) of the original signal input from the input terminal 1, and the waveform indicated by the broken line is the distortion waveform (output waveform) generated by the distortion waveform generation unit 2. In FIG. 2, the horizontal axis indicates time, and the vertical axis indicates the level of signal amplitude. For example, in the section shown as “section P1” in FIG. 2, the magnitude of the original signal indicated by the solid line increases with time, and the slope of the waveform is upward. Therefore, the distortion waveform generation unit 2 operates as an APF. Thereby, the waveform that follows the waveform of the input original signal is output from the distortion waveform generation unit 2.

  On the other hand, in the section indicated as “P2”, the magnitude of the original signal indicated by the solid line decreases with time, and the slope of the waveform is downward. Therefore, the distortion waveform generation unit 2 operates as an LPF. As a result, the distortion waveform generation unit 2 outputs a signal in which a predetermined high-frequency component is cut from the input original signal. The waveform obtained through the LPF falls with a gentler slope than the slope of the waveform of the original signal.

  By performing such processing, the slope of the distortion waveform generated by the distortion waveform generation unit 2 changes when the section P1 to which the APF is applied is switched to the section P2 to which the LPF is applied. Since the section obtained by adding the section P1 to which the APF is applied and the section P2 to which the LPF is applied substantially coincides with one period of the frequency of the input signal, different transfer characteristics are applied depending on the change direction of the slope of the waveform. The generated signal is generated and the harmonics of the original signal are generated. That is, in the present embodiment, by switching between APF and LPF according to the slope of the waveform of the input original signal, a distorted wave of the waveform of the input signal is generated and a harmonic of the input signal is generated. Yes.

  As in the conventional case, when the input signal is distorted by applying an absolute value circuit, or when the peak value held as described in Patent Document 1 is reset at the zero cross point, the generated distorted wave and the input signal Correlation with the waveform becomes low. On the other hand, in the present embodiment, a distorted wave is generated by switching between APF and LPF. Therefore, the generated distorted wave shows a frequency response corresponding to the input signal.

  Further, in the section P2 to which the LPF is applied, an envelope-shaped waveform having a waveform with a gentler slope than the slope of the waveform of the original signal is generated. That is, a waveform from which high frequency components included in the original signal are removed is generated. Accordingly, since no extra harmonics are generated in this section, it is possible to easily and accurately generate low-frequency harmonics to be interpolated.

  A signal generated by the distortion waveform generation unit 2 and from which a predetermined high-frequency component (component other than the harmonics of the acoustic signal) is removed by the BPF 3 is added to the original signal by the adder 5 and output to the output terminal 6. . The signal output to the output terminal 6 is a signal in which the harmonics of the original signal are superimposed on the original signal. That is, the output terminal 6 outputs a signal obtained by interpolating a signal of a predetermined band.

  FIG. 3 is a graph in which the signal after passing through the BPF 3 and the original signal input to the input terminal 1 are displayed together. The waveform indicated by the thin line in FIG. 3 is the waveform of the original signal input from the input terminal 1, and the waveform indicated by the thick line is the waveform of the signal that has passed through the BPF 3. In FIG. 3, the horizontal axis indicates the frequency (Hz), and the vertical axis indicates the signal magnitude (dB). In the input signal indicated by a thin line, the magnitude of -30 dB to -80 dB between 0 Hz and 100 Hz decreases rapidly to about -130 dB at 100 Hz or higher. That is, it can be seen that the input signal is a sound source in which a signal having a frequency of 100 MHz or more is cut. Moreover, between 100 Hz and 200 Hz, the frequency of the band extracted by the BPF 3 from the harmonic signal generated by the distortion waveform generation unit 2 is output.

Then, the adder 5 adds these two signals, so that the input signal combines a thick line and a thin line, and a signal in a band from 100 Hz to 200 Hz is added and generated. That is, this signal is an enhanced signal. The input signal is a sound source up to 100 Hz as shown by a thin line in FIG. 3. For example, if the speaker cannot reproduce 100 Hz or less, the sound cannot be heard. However, by adding (adding) the output of the BPF 3 (harmonics from 100 Hz to 200 Hz) indicated by a bold line in FIG. 3, it is possible to listen to the sound through the speaker.

  Particularly in high-frequency interpolation, it is known that the generated speech becomes more natural by emphasizing a portion where the change in the amplitude of the input signal is severe by interpolation. According to this embodiment, Also, high-frequency interpolation can be performed well. In this embodiment, since the LPF functions from a position where the slope of the waveform of the input signal changes, the generated distortion waveform becomes a gentle downward envelope waveform at the change point. As a result, the obtained harmonics are not limited to odd multiples, as in the case of obtaining a rectangular wave by peak hold. That is, since it is possible to obtain harmonics of fine multiples such as double, triple, and quadruple, a finer sound can be obtained.

[2. Example of configuration and processing of distortion waveform generation unit]
Next, an example of the configuration and processing of the distortion waveform generation unit 2 according to this embodiment will be described with reference to FIGS. 4 and 5. FIG. 4 is a block diagram illustrating a configuration example of the distortion waveform generation unit 2. In the present embodiment, the distortion waveform generation unit 2 is composed of a primary IIR filter. The order of the IIR filter is not limited to the first order, but may be calibrated with other orders such as the second order. The distortion waveform generator 2 includes an adder 21 as a first adder, a multiplier 22 as a first multiplier, a multiplier 23 as a second multiplier, a delay element 24, a third element And a multiplier 26 as a fourth multiplier. In addition, the distortion waveform generation unit 2 includes an adder 27 as a second adder, a comparator 28, and a selection unit 29.

  The adder 21 adds the input signal of the value X input to the distortion waveform generation unit 2 and the output signal from the multiplier 22 in which the coefficient a1 as the first coefficient is set, and outputs the result. The multiplier 23 supplies a signal obtained by multiplying the input signal by c 1 to the delay element 24 and / or the multiplier 22 and the multiplier 26. The delay element 24 adds a delay of one period to the input signal and outputs it. A signal output from the adder 21 or a signal output from the multiplier 23 is input to the delay element 24.

  The selection unit 29 includes a switch 291 that switches a signal input to the delay element 24 and a switch 292 that switches a signal input to the multiplier 22 and the multiplier 26. The switch 291 has a contact C1a connected to the transmission line of the signal output from the multiplier 23, a contact C1b connected to the transmission line of the signal output from the adder 21, and a common connected to the delay element 24. A contact C1c. The switch 292 is connected to a contact C2a connected to the transmission line of the signal output from the multiplier 23, a contact C2b connected to the delay element 24, and a transmission line that supplies signals to the multiplier 22 and the multiplier 26. Common contact C2c.

  The multiplier 22 multiplies the input signal by the coefficient a1 and outputs the result to the adder 21. Multiplier 26 multiplies the input signal by coefficient b 1 and outputs the result to adder 27. The multiplier 25 is provided at the output destination of the adder 21, and multiplies the signal output from the adder 21 by the coefficient b 1 and outputs the result to the adder 27. The adder 27 adds the signal output from the multiplier 25 and the signal output from the multiplier 26 and outputs the result as an output signal Y, and also inputs the output signal Y to the comparator 28.

  The comparator 28 compares the value X of the input signal with the value Y of the output signal, and switches the selection destination of the selection unit 29 according to the comparison result. More specifically, when X ≧ Y, the common contact C1c of the switch 291 is connected to the contact C1a, and the common contact C2c of the switch 292 is connected to the contact C2a. That is, the signal output from the multiplier 23 is selected as a signal to be input to the delay element 24, and the signal output from the multiplier 23 is selected as a signal to be input to the multiplier 22 and the multiplier 26. On the other hand, when X <Y, the common contact C1c of the switch 292 is connected to the contact C1b, and the common contact C2c of the switch 292 is connected to the contact C2b. That is, the signal output from the adder 21 is selected as a signal to be input to the delay element 24, and the signal output from the delay element 24 is selected as a signal to be input to the multiplier 22 and the multiplier 26.

  FIG. 5 is a diagram for easily explaining the function of the distortion waveform generation unit 2 that switches according to the comparison result of the comparator 28. In FIG. 5, portions corresponding to those in FIG. 4 are denoted with the same reference numerals, and redundant description is omitted. FIG. 5A shows the configuration of a filter that operates when X ≧ Y, and FIG. 5B shows the configuration of a filter that operates when X <Y.

The configuration shown in FIG. 5A is the same as that of a general first-order IIR filter (LPF). Here, the coefficients a1 and b1 are values calculated from the cut-off frequency of the LPF by general conversion, for example, SZ conversion. In the delay element 24, the value X of the input signal is held for the first time, and thereafter, a value obtained by adding the input signal X, which is delayed by the delay element 24 and multiplied by the coefficient a1, and the input signal X is obtained. Retained. Assuming that the signal delayed by the delay element 24 is “d”, the output signal Y from the LPF can be calculated by the following Equation 1.
Y = (X + d × coefficient a1) × coefficient b1 + d × coefficient b1 Equation 1
That is, when X <Y, that is, when the slope of the waveform of the original signal input to the distortion waveform generation unit 2 is downward, a predetermined high frequency is determined according to the filter characteristics determined by the values of the coefficients a1 and b1. Ingredients are cut.

On the other hand, in the configuration of the filter that operates when X ≧ Y shown in FIG. 5B, the multiplier 23 and the multiplier 26, and the delay element 24, the coefficient c 1 is added to the input signal X by the multiplier 23. The multiplied value is supplied. That is, X × coefficient c1 is substituted into “d” shown in the above equation 1. The following Expression 2 is an expression in which “d” is not used in place of “d” in Expression 1.
Y = (X + X × coefficient c1 × coefficient a1) × coefficient b1 + coefficient c1 × coefficient b1 Equation 2
Here, when X = Y, the coefficient c1 is expressed by the following expression 3.
Coefficient c1 = (1−coefficient b1) / (coefficient a1 × coefficient b1 + coefficient b1) Equation 3
That is, the distortion waveform generator 2 is configured as shown in FIG. 5 and the coefficient c1 is set to the value shown in Equation 3, so that the output signal Y can follow the input signal X. That is, the distortion waveform generation unit 2 can function as an APF.

  Each time multiplication is performed by the multiplier 23, the value of the result is input to the delay element 24, and the value at that time is held in the delay element 24. As a result, the function of charging the voltage to the capacitor in the LPF of the analog circuit is realized.

  By configuring the distortion waveform generation unit 2 in this way and performing the calculation, it is possible to obtain the output signal Y with a very simple calculation and a very small number of multiplications of about four times.

  In this embodiment, as described above, when the value X of the input signal is equal to or larger than the value Y of the output signal (X ≧ Y), APF is applied, and the value X of the input signal is changed to the output signal. If the value is smaller than Y (X <Y), LPF is applied. Thereby, when the slope of the waveform of the input signal is upward, a waveform that follows the waveform of the input signal is output, and when it is downward, the waveform becomes a gentle downward envelope. That is, a distorted waveform having a waveform corresponding to the waveform of the input signal is generated.

  Here, a case where a distorted waveform having a waveform corresponding to the waveform of the input signal is generated by another method is compared with a case where a distorted waveform is generated by the method according to the present embodiment. As another method, consider an example in which a distortion waveform of an input signal is detected by peak hold processing. As a means for realizing the peak hold processing by digital signal processing, “Hilbert transform” is generally well known.

  FIG. 6 is a diagram showing a distortion waveform of the input signal detected using the Hilbert transform superimposed on the waveform of the input signal. The waveform shown by the solid line in FIG. 6 is the waveform of the input signal (input waveform), and the waveform of the distortion waveform (output waveform) generated by the waveform shown by the broken line. In FIG. 6, the horizontal axis indicates time, and the vertical axis indicates the level of signal amplitude.

  As shown in FIG. 6, it can be seen that the distortion waveform generated using the Hilbert transform also follows the high frequency component of the input signal. This high frequency component includes a component that is not necessary for the interpolation of the acoustic signal, and when such an unnecessary high frequency component is used for the interpolation of the acoustic signal, the portion is different. Perceived as sound. That is, the sound quality of the reproduced voice is deteriorated.

  On the other hand, as shown in FIG. 7, in the distortion waveform detected by the method according to the present embodiment, a high frequency component that appears in a portion surrounded by an ellipse in a section where a downward envelope is output. Is removed. As a result, an excessively high frequency component is not included in the interpolation target of the acoustic signal, and only the harmonics necessary for the interpolation of the acoustic signal can be easily and efficiently extracted.

  Further, in the present embodiment, a signal in which filters having different transfer characteristics are alternately applied is generated by switching between APF and LPF according to the slope of the waveform of the input signal. Therefore, when this waveform is downward, a distorted waveform indicating a frequency response corresponding to the waveform of the input signal is output even at the enveloped portion. For example, it can be seen that in two places surrounded by a broken-line ellipse in FIG. 7, the amplitude of the corresponding input signal is increased, and the distorted waveform is a waveform linked to it.

  FIG. 7 is a graph showing a distorted waveform when the frequency of the input signal is low. Even when a high-frequency input signal is input, the distorted waveform showing a frequency response according to the waveform of the input signal. Is output. FIG. 8 shows a diagram in which a distortion waveform (output waveform) generated from an input signal having a high frequency is superimposed on the waveform (input waveform) of the input signal. The waveform indicated by the solid line in FIG. 8 is the waveform (input waveform) of the original signal input from the input terminal 1, and the waveform indicated by the broken line is the distortion waveform (output waveform) generated by the distortion waveform generation unit 2. The horizontal axis in FIG. 2 indicates time, and the vertical axis indicates the level of signal amplitude. Even in the case of processing a signal having a high frequency as shown in FIG. 8, the envelope follows the input signal in a section where the value of the input signal is increasing, and gently envelopes downward in a section where the value of the input signal decreases. It can be seen that a distorted waveform is generated.

  According to the above-described embodiment, the harmonics of the input signal can be generated by a simple method of switching between APF and LPF according to the slope of the waveform of the input signal. Since the harmonics generated at this time are obtained by filtering the input signal, the frequency response corresponding to the input signal is shown. Therefore, the signal extracted from the harmonics generated in this way is added to the original signal, thereby realizing a more natural interpolation of the acoustic signal.

  Further, according to the above-described embodiment, in the section where the downward envelope is output, the high frequency component of the original signal is removed, so that an extra high frequency component is added to the target of the acoustic signal interpolation. It will not be included. That is, only the harmonics necessary for the interpolation of the acoustic signal can be easily and efficiently extracted.

  Further, according to the above-described embodiment, the distortion waveform of the input signal can be generated with a very simple logic using the first-order IIR filter, so that the calculation amount is an extremely small amount of about four times. It can be. On the other hand, for example, when the distortion waveform of the input signal is detected by the above-described Hilbert transform, about 80th order calculation is required to obtain the same result as that obtained by the present invention. This is because the Hilbert transform is a filter based on the FIR filter.

  For this reason, particularly in a small-sized device, when the processing circuit (CPU) is used in combination with other functions (video display or the like) performed by the device, there arises a problem that the load on the processing circuit increases. In addition, the power consumption increases as the load on the processing circuit increases. Furthermore, manufacturing costs increase by taking measures to solve such problems. According to the present embodiment, since harmonics for acoustic signal interpolation can be generated with a very small amount of calculation, it is possible to reduce manufacturing costs and power consumption, and it is easy to mount on embedded devices.

  When the distortion waveform generation unit 2 is configured by a DSP as in the present embodiment, the above-described distortion waveform generation processing can be realized on software, but the present invention is not limited to this. . The processing circuit for each distortion waveform calculation step described above may be configured by hardware (analog circuit).

DESCRIPTION OF SYMBOLS 1 ... Input terminal, 2 ... Distortion waveform generation part, 3 ... Band pass filter (BPF), 4 ... Delay circuit, 5 ... Adder, 6 ... Output terminal, 10 ... Acoustic signal processing apparatus, 21 ... Adder, 22, 23 ... Multiplier, 24 ... Delay element, 25,26 ... Multiplier, 27 ... Adder, 28 ... Comparator, 29 ... Selection unit

Claims (7)

During the period when the slope of the waveform of the original signal supplied to the input terminal is upward, a waveform that follows the waveform of the original signal is output, and during the period when the slope of the waveform of the original signal is downward, A distortion waveform generator that generates a distortion waveform by outputting a waveform of a signal that does not include a predetermined high frequency of
A harmonic extraction unit that extracts a signal of a predetermined frequency band as a harmonic of the original signal from the distortion waveform generated by the distortion waveform generation unit;
An acoustic signal processing apparatus comprising: an adder that adds the harmonics extracted by the harmonic extraction unit to an original signal supplied to the input terminal. The distortion waveform generation unit
The acoustic signal processing device according to claim 1, comprising filter characteristics of both an all-pass filter and a low-pass filter. The acoustic signal processing device according to claim 2, wherein the low-pass filter includes an IIR filter. The all-pass filter is configured by adding, to the IIR filter, a multiplier in which a coefficient for causing the value of the output signal from the distortion waveform generation unit to follow the value of the input signal to the distortion waveform generation unit is set. Item 4. The acoustic signal processing device according to Item 3. The distortion waveform generation unit
A first adder that adds an input signal input to the distortion waveform generation unit and a signal output from a first multiplier that multiplies the input signal by a first coefficient;
A second multiplier that multiplies an input signal input to the distortion waveform generation unit by a second coefficient;
A delay element that adds a predetermined delay to the signal output from the second multiplier or the signal output from the first adder;
A third multiplier for multiplying the signal output from the first adder by a third coefficient;
A fourth multiplier that multiplies a signal output from the first adder and delayed by the delay element, or a signal output from the second multiplier by a third coefficient;
A second adder for adding the signal output from the fourth multiplier and the signal output from the third multiplier; a value of the output signal output from the second adder; A comparator that compares the value of the input signal input to the distortion waveform generator;
Either one of the signal output from the delay element and the signal output from the second multiplier is selected, and both the first multiplier and the fourth multiplier are selected. The acoustic signal processing apparatus according to claim 3, further comprising: a selection unit that outputs. When the comparator determines that the value of the input signal is greater than or equal to the value of the output signal, the selection unit selects the signal output from the second multiplier, and the second multiplier The signal output from the delay element is stored in the delay element, and when the comparator determines that the value of the input signal is less than the value of the output signal, the selection unit outputs the signal from the delay element. Selected signal,
The second coefficient set in the second multiplier is expressed by the following formula: second coefficient = (1−third coefficient) / (first coefficient × third coefficient + second Coefficient of 3)
The acoustic signal processing apparatus according to claim 3. During the period when the slope of the waveform of the original signal supplied to the input terminal is upward, a waveform that follows the waveform of the original signal is output, and during the period when the slope of the waveform of the original signal is downward, Generating a distorted waveform by outputting a waveform of a signal that does not include a predetermined high frequency of
Extracting a signal of a predetermined frequency band from the distortion waveform;
Adding the extracted signal to an original signal supplied to the input terminal.