Digital sound processing device, digital sound processing method and digital sound processing program

Abstract

The even harmonic adding section (171) adds a correction value (Vadd) to a first adjacent sample (S1) next to a first minimum sample (S0), and subtracts the correction value (Vsub) from a second adjacent sample (S2) of a sample preceding a maximum sample (S3). The even harmonic adding section (171) adds a correction value (Vadd) to a third adjacent sample (S4) next to a maximum sample (S3), and subtracts the correction value (Vsub) from a fourth adjacent sample (S5) of a sample preceding a second minimum sample (S6). The odd harmonic adding section (172) subtracts the correction value (Vsub) from the first minimum sample (S0) and the second minimum sample (S6), and adds the correction value (Vadd) to the maximum sample (S3).

Description

Technical Field

[0001] The present invention relates to a digital sound processing device, a digital sound processing method and a digital sound processing program for processing digital sound signals. Background Art

[0002] Patent document 1 describes a digital sound processing device that processes a digital sound signal to improve sound quality as follows. The digital sound processing device detects the sampling interval between a maximum sample having a maximum value and a minimum sample having a minimum value in the waveform of the digital sound signal. The digital sound processing device adds a correction value obtained by multiplying the difference between the maximum sample and the adjacent sample by a coefficient less than 1 to the adjacent samples before and after the maximum sample, and subtracts a correction value obtained by multiplying the difference between the minimum sample and the adjacent sample by a coefficient less than 1 from the adjacent samples before and after the minimum sample.

[0003] Prior art literature

[0004] Patent Literature

[0005] Patent document 1: Japanese Patent No. 3401171;

[0006] Patent document 2: Japanese Patent No. 3659489;

[0007] Patent document 3: Japanese Patent No. 4985570. Summary of the invention

[0008] According to the digital sound processing device described in Patent Document 1, only odd harmonics are added to the digital sound signal. In the digital sound processing device described in Patent Document 2, only odd harmonics are added to the digital sound signal. The odd harmonics added to the fundamental tone basically improve the sound quality, but sometimes they may be irritating and make people feel uncomfortable depending on the person. Generally, most people usually feel comfortable with the even harmonics added to the fundamental tone. Therefore, it is desirable to add both even harmonics and odd harmonics to the digital sound signal.

[0009] Patent document 3 describes a structure for adding both even harmonics and odd harmonics to a digital sound signal. The digital sound processing device described in Patent document 3 adds or subtracts a correction value to only one of the waveform portion where the sampling value rises from a minimum sampling value to a maximum sampling value and the waveform portion where the sampling value falls from a maximum sampling value to a minimum sampling value. According to this structure, both even harmonics and odd harmonics are added to a digital sound signal.

[0010] However, it is not ideal to correct only one of the waveforms of the rising and falling waveforms. It is desirable to correct both the waveforms of the rising and falling waveforms and add both even harmonics and odd harmonics to the digital sound signal.

[0011] The purpose of the implementation is to provide a digital sound processing device, a digital sound processing method and a digital sound processing program, which can correct the waveform of both the waveform part where the sampling value rises and the waveform part where the sampling value falls, and add both even harmonics and odd harmonics to the digital sound signal.

[0012] According to a first embodiment, a digital sound processing device is provided, comprising: a maximum sampling detection unit, which calculates the maximum value in the samples constituting the input digital sound signal, and detects the maximum sampling with the maximum value; a minimum sampling detection unit, which calculates the minimum value in the samples constituting the digital sound signal, and detects the minimum sampling with the minimum value; a waveform tilt determination unit, which determines whether the samples constituting the digital sound signal are the waveform part in which the sampling value rises from the minimum sampling to the maximum sampling, or the waveform part in which the sampling value falls from the maximum sampling to the minimum sampling; a counter, which counts the sampling intervals between the minimum sampling and the maximum sampling adjacent in the time direction; a coefficient selection unit, which selects the coefficients for even harmonics and the coefficients for odd harmonics according to the sampling interval counted by the counter; and a higher harmonic component adding unit, which has an even harmonic adding unit for adding even harmonics to the digital sound signal and an odd harmonic adding unit for adding odd harmonics, and adds higher harmonic components consisting of even harmonics and odd harmonics to the digital sound signal and outputs it.

[0013] In the above-mentioned digital sound processing device, when the samples constituting the digital sound signal rise from a first minimum sample to a first maximum sample and fall from the first maximum sample to a second minimum sample, the even harmonic adding unit performs any one of a first even harmonic adding process and a second even harmonic adding process.

[0014] In the above-mentioned digital sound processing device, as the first even harmonic addition processing, the even harmonic addition unit adds even harmonics to the digital sound signal by: adding a first correction value to a first adjacent sample, the first adjacent sample is the next sample after the first minimum sample, the first correction value is obtained by multiplying the first difference between the first minimum sample and the first adjacent sample by a first even harmonic coefficient, the first even harmonic coefficient is selected by the coefficient selection unit according to the first sampling interval between the first minimum sample and the first maximum sample; subtracting a second correction value from a second adjacent sample, the second adjacent sample is the sample before the first maximum sample, the second correction value is the difference between the second adjacent sample and the first maximum sample. The second difference between the first maximum samples is multiplied by the first even harmonic coefficient; a third correction value is added to the third adjacent sample, the third adjacent sample is the next sample after the first maximum sample, the third correction value is obtained by multiplying the third difference between the first maximum sample and the third adjacent sample by the second even harmonic coefficient, the second even harmonic coefficient is selected by the coefficient selection unit according to the second sampling interval between the first maximum sample and the second minimum sample; and a fourth correction value is subtracted from the fourth adjacent sample, the fourth adjacent sample is the previous sample of the second minimum sample, and the fourth correction value is obtained by multiplying the fourth difference between the fourth adjacent sample and the second minimum sample by the second even harmonic coefficient.

[0015] In the above-mentioned digital sound processing device, as the second even harmonic addition processing, the even harmonic adding unit adds even harmonics to the digital sound signal by: subtracting the first correction value from the first adjacent samples; adding the second correction value to the second adjacent samples; subtracting the third correction value from the third adjacent samples; and adding the fourth correction value to the fourth adjacent samples.

[0016] In the above-mentioned digital sound processing device, the odd harmonic adding unit performs odd harmonic adding processing of adding odd harmonics to the digital sound signal by: subtracting a fifth correction value from the first minimum sample, the fifth correction value being obtained by multiplying a fifth difference between the first minimum sample and a fifth adjacent sample by a first odd harmonic coefficient, the fifth adjacent sample being a sample before the first minimum sample, the first odd harmonic coefficient being selected by the coefficient selecting unit according to a third sampling interval, the third sampling interval being an interval between the first minimum sample and a second maximum sample immediately before the first minimum sample; adding a sixth correction value to the first maximum sample, the sixth correction value being obtained by multiplying the second difference by a second odd harmonic coefficient, the second odd harmonic coefficient being selected by the coefficient selecting unit according to the first sampling interval; and subtracting a seventh correction value from the second minimum sample, the seventh correction value being obtained by multiplying the fourth difference by a third odd harmonic coefficient, the third odd harmonic coefficient being selected by the coefficient selecting unit according to the second sampling interval.

[0017] According to a second embodiment, a digital sound processing method is provided, which calculates the maximum value in the samples constituting the input digital sound signal and detects the maximum sample with the maximum value; calculates the minimum value in the samples constituting the digital sound signal and detects the minimum sample with the minimum value; determines whether the samples constituting the digital sound signal are the waveform part where the sampling value rises from the minimum sample to the maximum sample, or the waveform part where the sampling value falls from the maximum sample to the minimum sample; and counts the sampling intervals between the minimum sample and the maximum sample adjacent in the time direction.

[0018] In the above-mentioned digital sound processing method, when the samples constituting the digital sound signal rise from a first minimum sample to a first maximum sample and fall from the first maximum sample to a second minimum sample, any one of the first even harmonic addition processing and the second even harmonic addition processing is performed.

[0019] In the above-mentioned digital sound processing method, as the first even harmonic addition processing, the digital sound signal is added with even harmonics by: adding a first correction value to a first adjacent sample, the first adjacent sample is the next sample after the first minimum sample, the first correction value is obtained by multiplying a first difference between the first minimum sample and the first adjacent sample by a first even harmonic coefficient, the first even harmonic coefficient is selected according to a first sampling interval between the first minimum sample and the first maximum sample; subtracting a second correction value from a second adjacent sample, the second adjacent sample is the sample before the first maximum sample, the second correction value is the difference between the second adjacent sample and the first maximum sample The third correction value is added to the third adjacent sample, the third adjacent sample is the next sample after the first maximum sample, the third correction value is obtained by multiplying the third difference between the first maximum sample and the third adjacent sample by the second even harmonic coefficient, and the second even harmonic coefficient is selected according to the second sampling interval between the first maximum sample and the second minimum sample; and the fourth correction value is subtracted from the fourth adjacent sample, the fourth adjacent sample is the previous sample of the second minimum sample, and the fourth correction value is obtained by multiplying the fourth difference between the fourth adjacent sample and the second minimum sample by the second even harmonic coefficient.

[0020] In the above-mentioned digital sound processing method, as the second even harmonic addition processing, even harmonics are added to the digital sound signal by: subtracting the first correction value from the first adjacent samples; adding the second correction value to the second adjacent samples; subtracting the third correction value from the third adjacent samples; and adding the fourth correction value to the fourth adjacent samples.

[0021] In the above-mentioned digital sound processing method, odd harmonic addition processing of adding odd harmonics to the digital sound signal is performed by: subtracting a fifth correction value from the first minimum sample, the fifth correction value being obtained by multiplying a fifth difference between the first minimum sample and a fifth adjacent sample by a first odd harmonic coefficient, the fifth adjacent sample being a sample before the first minimum sample, the first odd harmonic coefficient being selected according to a third sampling interval, the third sampling interval being an interval between the first minimum sample and a second maximum sample immediately before the first minimum sample; adding a sixth correction value to the first maximum sample, the sixth correction value being obtained by multiplying the second difference by a second odd harmonic coefficient, the second odd harmonic coefficient being selected according to the first sampling interval; and subtracting a seventh correction value from the second minimum sample, the seventh correction value being obtained by multiplying the fourth difference by a third odd harmonic coefficient, the third odd harmonic coefficient being selected according to the second sampling interval.

[0022] According to a third embodiment of the present invention, a digital sound processing program is provided, so that a computer executes the following steps: calculating the maximum value in the samples constituting the input digital sound signal, and detecting the maximum sample with the maximum value; calculating the minimum value in the samples constituting the digital sound signal, and detecting the minimum sample with the minimum value; determining whether the samples constituting the digital sound signal are the waveform part in which the sampling value rises from the minimum sample to the maximum sample, or the waveform part in which the sampling value falls from the maximum sample to the minimum sample; and counting the sampling intervals between the minimum sample and the maximum sample adjacent in the time direction.

[0023] In the above-mentioned digital sound processing program, when the samples constituting the digital sound signal rise from a first minimum sample to a first maximum sample and fall from the first maximum sample to a second minimum sample, any one of the first even harmonic adding step and the second even harmonic adding step is executed.

[0024] In the above-mentioned digital sound processing program, as the first even harmonic adding step, even harmonics are added to the digital sound signal, so that the computer executes the following steps: adding a first correction value to the first adjacent sample, the first adjacent sample is the next sample after the first minimum sample, the first correction value is obtained by multiplying the first difference between the first minimum sample and the first adjacent sample by the first even harmonic coefficient, and the first even harmonic coefficient is selected according to the first sampling interval between the first minimum sample and the first maximum sample; subtracting a second correction value from the second adjacent sample, the second adjacent sample is the sample before the first maximum sample, and the second correction value is the difference between the second adjacent sample and the first maximum sample. The second difference between the largest samples is multiplied by the first even harmonic coefficient; a third correction value is added to the third adjacent sample, the third adjacent sample is the next sample after the first maximum sample, the third correction value is obtained by multiplying the third difference between the first maximum sample and the third adjacent sample by the second even harmonic coefficient, and the second even harmonic coefficient is selected according to the second sampling interval between the first maximum sample and the second minimum sample; and a fourth correction value is subtracted from the fourth adjacent sample, the fourth adjacent sample is the previous sample of the second minimum sample, and the fourth correction value is obtained by multiplying the fourth difference between the fourth adjacent sample and the second minimum sample by the second even harmonic coefficient.

[0025] In the above-mentioned digital sound processing program, as a second even harmonic addition step, the computer executes the following steps: subtracting the first correction value from the first adjacent samples; adding the second correction value to the second adjacent samples; subtracting the third correction value from the third adjacent samples; and adding the fourth correction value to the fourth adjacent samples.

[0026] The digital sound processing program enables the computer to execute an odd harmonic adding step, which includes the following steps: subtracting a fifth correction value from the first minimum sample, the fifth correction value being obtained by multiplying a fifth difference between the first minimum sample and a fifth adjacent sample by a first odd harmonic coefficient, the fifth adjacent sample being a sample preceding the first minimum sample, the first odd harmonic coefficient being selected according to a third sampling interval, the third sampling interval being an interval between the first minimum sample and a second maximum sample immediately preceding the first minimum sample; adding a sixth correction value to the first maximum sample, the sixth correction value being obtained by multiplying the second difference by a second odd harmonic coefficient, the second odd harmonic coefficient being selected according to the first sampling interval; and subtracting a seventh correction value from the second minimum sample, the seventh correction value being obtained by multiplying the fourth difference by a third odd harmonic coefficient, the third odd harmonic coefficient being selected according to the second sampling interval.

[0027] According to the digital sound processing device, digital sound processing method and digital sound processing program of the embodiment, it is possible to correct the waveforms of both the waveform part where the sampling value rises and the waveform part where the sampling value falls, and add both even harmonics and odd harmonics to the digital sound signal. BRIEF DESCRIPTION OF THE DRAWINGS

[0028] Figure 1 This is a block diagram showing a digital sound processing device according to one embodiment.

[0029] Figure 2 This is a waveform diagram showing an example of a digital audio signal input to a digital audio processing device according to one embodiment.

[0030] Figure 3 1 is a diagram showing an example of a coefficient table indicating coefficients set for each sampling interval between a maximum sample and a minimum sample.

[0031] Figure 4 3 is a waveform diagram showing the first even harmonic addition process and the odd harmonic addition process when the sampling interval is 3 fs.

[0032] Figure 5 3 is a waveform diagram showing the second even harmonic addition process and odd harmonic addition process when the sampling interval is 3 fs.

[0033] Figure 6 3 is a waveform diagram showing the first even harmonic addition process and the odd harmonic addition process when the sampling interval is 6 fs.

[0034] Figure 7 3 is a waveform diagram showing the second even harmonic addition process and the odd harmonic addition process when the sampling interval is 6 fs.

[0035] Figure 8 3 is a waveform diagram showing the first even harmonic addition process and the odd harmonic addition process when the sampling interval is 2fs.

[0036] Fig. 9 3 is a waveform diagram showing the second even harmonic addition process and odd harmonic addition process when the sampling interval is 2fs.

[0037] Fig.10 The present invention is a flowchart showing a process executed by a digital sound processing device according to one embodiment, a digital sound processing method according to one embodiment, and a process of causing a computer to execute a digital sound processing program according to one embodiment.

[0038] Fig.11A This is a waveform diagram showing an example of a waveform with a sampling interval of 3fs in which an inversion phenomenon occurs due to correction of a digital audio signal.

[0039] Fig. 11B It is shown by Fig.11A The waveform diagram shown is a waveform diagram of a state where an inversion phenomenon occurs due to waveform correction with a sampling interval of 3 fs.

[0040] Fig. 12A This is a waveform diagram showing an example of a waveform with a sampling interval of 4fs in which an inversion phenomenon occurs due to correction of a digital audio signal.

[0041] Fig. 12B It is shown by Fig. 12A The waveform diagram shown is a waveform diagram of a state where an inversion phenomenon occurs due to waveform correction with a sampling interval of 4 fs.

[0042] Fig.13 This is a flowchart showing a first example of processing for avoiding the inversion phenomenon in the case of a waveform with a sampling interval of 3 fs.

[0043] Fig.14 This is a flowchart showing a second example of processing for avoiding the inversion phenomenon in the case of a waveform with a sampling interval of 3 fs.

[0044] Fig.15A It is shown by Fig.13 or Fig.14 The waveform diagram shown is a first example of a state in which the process avoids the inversion phenomenon.

[0045] Fig. 15B It is shown by Fig.13 or Fig.14 The waveform diagram shown is a second example of a state in which the inversion phenomenon is avoided.

[0046] Fig.16This is a flowchart showing a process for avoiding the inversion phenomenon in the case of a waveform with a sampling interval of 4 fs or more.

[0047] Fig.17A It is shown by Fig.16 The waveform diagram shown is a first example of a state in which the process avoids the inversion phenomenon.

[0048] Fig. 17B It is shown by Fig.16 The waveform diagram shown is a second example of a state in which the inversion phenomenon is avoided.

[0049] Fig.18 This is a block diagram showing a configuration example of a microcomputer that executes a digital sound processing program according to one embodiment. DETAILED DESCRIPTION

[0050] Hereinafter, a digital sound processing device, a digital sound processing method, and a digital sound processing program according to an embodiment will be described with reference to the drawings.

[0051] exist Figure 1 In one embodiment, a digital sound processing device 100 includes: a maximum sampling detection unit 11, a minimum sampling detection unit 12, a waveform tilt determination unit 13, a counter 14, a coefficient selection unit 15, a coefficient table holding unit 16, and a higher harmonic component adding unit 17. The higher harmonic component adding unit 17 includes an even harmonic adding unit 171 and an odd harmonic adding unit 172. The digital sound processing device 100 can be composed of hardware including a circuit, can be composed of software, or can be composed of a mixture of hardware and software. The digital sound processing device 100 can also be composed of an integrated circuit.

[0052] A digital audio signal having a predetermined number of quantization bits and a predetermined sampling frequency is input to the maximum sampling detection unit 11 and the minimum sampling detection unit 12. The maximum sampling detection unit 11 detects a maximum sampling having a maximum value by determining the magnitude relationship between adjacent samples in the input digital audio signal. The minimum sampling detection unit 12 similarly detects a minimum sampling having a minimum value. The maximum sampling and the minimum sampling are provided to the waveform tilt determination unit 13 and the counter 14.

[0053] The waveform tilt determination unit 13 determines whether it is a waveform portion where the sampling value rises or a waveform portion where the sampling value falls, based on the order in which the maximum sampling and the minimum sampling are input. If the maximum sampling is input after the minimum sampling, the waveform tilt determination unit 13 determines that the waveform portion between the minimum sampling and the maximum sampling is a waveform portion where the sampling value rises. If the minimum sampling is input after the maximum sampling, the waveform tilt determination unit 13 determines that the waveform portion between the maximum sampling and the minimum sampling is a waveform portion where the sampling value falls. The determination result of the waveform tilt determination unit 13 is provided to the higher harmonic component adding unit 17.

[0054] The counter 14 detects the sampling intervals between the extremely small samples and the extremely large samples. Figure 2 FIG. 2 shows an example of a waveform of a digital audio signal input to the digital audio processing device 100. Figure 2 In the example, sampling S0 is a very small sampling and sampling S3 is a very large sampling. Figure 2 The sampling interval between the minimum sample S0 and the maximum sample S3 shown is 3. If the interval between adjacent minimum samples and maximum samples is recorded as fs, then Figure 2 The interval between the minimum sample S0 and the maximum sample S3 shown is 3fs.

[0055] The sampling intervals of the minimum and maximum samples include both the sampling intervals of the minimum and maximum samples in the waveform portion where the sampled value rises and the sampling intervals of the maximum and minimum samples in the waveform portion where the sampled value falls.

[0056] The sampling interval detected by the counter 14 is provided to the coefficient selection unit 15. The coefficient selection signal set by the user is input to the coefficient selection unit 15. The coefficient table holding unit 16 holds Figure 3 The coefficient table shown in FIG. 1 shows the coefficients used when generating correction values ​​to be added to or subtracted from the samples described later based on the sampling interval and the coefficient selection signal.

[0057] exist Figure 3 In , as the sampling interval, coefficients of 1 / 2 to 1 / 128 are set from 2 samples to 8 samples corresponding to the coefficient selection signals "00", "01", "10", and "11". Figure 3 The coefficients shown are examples and are not limited to Figure 3 The coefficients shown. The maximum sampling interval is not limited to 8 samples. The coefficient selection signals "00", "01", "10" and "11" are used as level selection signals for selecting the level of the correction value.

[0058] If the coefficient selection signal is set to "00", the degree of correction of the digital sound signal is the largest, and if the coefficient selection signal is set to "11", the degree of correction of the digital sound signal is the smallest. In addition, it is not necessary to adjust the correction value by selecting the coefficient according to the coefficient selection signal, but it is preferable to be able to adjust. In the case of not selecting the coefficient by the coefficient selection signal, any one of the coefficients of the coefficient selection signal "00", "01", "10", and "11" can be set in the coefficient table.

[0059] Figure 3The coefficients shown are the coefficients for even harmonics used when the even harmonic adding unit 171 adds even harmonics to the digital sound signal, and the coefficients for odd harmonics used when the odd harmonic adding unit 172 adds odd harmonics to the digital sound signal. The coefficients for even harmonics and the coefficients for odd harmonics may be the same coefficients or different coefficients.

[0060] return Figure 1 The coefficient selection unit 15 reads the coefficients for even harmonics used in the even harmonic adding unit 171 and the coefficients for odd harmonics used in the odd harmonic adding unit 172 from the coefficient table based on the sampling interval and the coefficient selection signal provided from the counter 14, and provides the coefficients for even harmonics and the coefficients for odd harmonics to the even harmonic adding unit 171 and the odd harmonic adding unit 172, respectively.

[0061] Here, use Figures 4 to 9 , the specific operations of the even harmonic adding unit 171 and the odd harmonic adding unit 172 are described. Figure 4 In the figure, the solid or dotted circles represent samples. The dotted lines represent samples before correction, and the solid lines represent samples after correction. The sampling interval between the minimum sample S0 and the maximum sample S3, and the sampling interval between the maximum sample S3 and the minimum sample S6 are both 3fs. It is assumed that the coefficient selection signal is set to "00".

[0062] The even harmonic adding unit 171 adds a correction value Vadd to the sample S1 after the very small sample S0, and the correction value Vadd is obtained by multiplying the difference between the very small sample S0 and the sample S1 by a coefficient. Sample S1 is corrected to sample S1' by adding the correction value Vadd. In addition, the even harmonic adding unit 171 subtracts a correction value Vsub from the sample S2 before the maximum sample S3, and the correction value Vsub is obtained by multiplying the difference between the sample S2 and the maximum sample S3 by a coefficient. Sample S2 is corrected to sample S2' by subtracting the correction value Vsub.

[0063] The correction value Vadd added to the sample S1 is 1 / 2 of the difference between the minimum sample S0 and the sample S1, and the correction value Vsub subtracted from the sample S2 is 1 / 2 of the difference between the sample S2 and the maximum sample S3.

[0064] Furthermore, the even harmonic adding unit 171 adds a correction value Vadd to the sample S4 following the maximum sample S3, and the correction value Vadd is obtained by multiplying the difference between the maximum sample S3 and the sample S4 by a coefficient. Sample S4 is corrected to sample S4' by adding the correction value Vadd. In addition, the even harmonic adding unit 171 subtracts a correction value Vsub from the sample S5 preceding the minimum sample S6, and the correction value Vsub is obtained by multiplying the difference between the sample S5 and the minimum sample S6 by a coefficient. Sample S5 is corrected to sample S5' by subtracting the correction value Vsub.

[0065] The correction value Vadd added to the sample S4 is 1 / 2 of the difference between the maximum sample S3 and the sample S4, and the correction value Vsub subtracted from the sample S5 is 1 / 2 of the difference between the sample S5 and the minimum sample S6.

[0066] The odd harmonic adding unit 172 subtracts a correction value Vsub from the minimum sample S0, which is obtained by multiplying the difference between the minimum sample S0 and the previous sample S99 by a coefficient. The minimum sample S0 is corrected to the minimum sample S0' by subtracting the correction value Vsub. If the sampling interval between the minimum sample S0 and the maximum sample adjacent to the front side of the minimum sample S0 is 3fs, the correction value Vsub subtracted from the minimum sample S0 is 1 / 2 of the difference between the sample S99 and the minimum sample S0.

[0067] Furthermore, the odd harmonic adding unit 172 adds a correction value Vadd to the maximum sample S3, which is obtained by multiplying the difference between the sample S2 and the maximum sample S3 by a coefficient. The maximum sample S3 is corrected to the maximum sample S3' by adding the correction value Vadd. The correction value Vadd added to the maximum sample S3 is 1 / 2 of the difference between the sample S2 and the maximum sample S3.

[0068] Furthermore, the odd harmonic adding unit 172 subtracts a correction value Vsub from the minimum sample S6, the correction value Vsub being obtained by multiplying the difference between the minimum sample S6 and the previous sample S5 by a coefficient. The minimum sample S6 is corrected to the minimum sample S6' by subtracting the correction value Vsub. The correction value Vsub subtracted from the minimum sample S6 is 1 / 2 of the difference between the minimum sample S6 and the sample S5.

[0069] The harmonic component adding unit 17 corrects the waveform shown by the single-dot chain line to the waveform shown by the solid line by adding the correction value Vadd or subtracting the correction value Vsub from the sample of the correction object of the digital sound signal as described above. The even harmonic adding unit 171 and the odd harmonic adding unit 172 also correct the digital sound signal in the same manner after sampling S7. Thus, harmonic components including even harmonics and odd harmonics are added to the digital sound signal.

[0070] like Figure 5 As shown, the harmonic component adding unit 17 may correct the digital audio signal by adding the correction value Vadd or subtracting the correction value Vsub from the sample to be corrected. Similarly, it is assumed that the coefficient selection signal is set to "00".

[0071] The even harmonic adding unit 171 subtracts the correction value Vsub from the sample S1, and the correction value Vsub is obtained by multiplying the difference between the minimum sample S0 and the sample S1 by a coefficient. The sample S1 is corrected to the sample S1' by subtracting the correction value Vsub. In addition, the even harmonic adding unit 171 adds the correction value Vadd to the sample S2, and the correction value Vadd is obtained by multiplying the difference between the sample S2 and the maximum sample S3 by a coefficient. The sample S2 is corrected to the sample S2' by adding the correction value Vadd.

[0072] The correction value Vsub subtracted from the sample S1 is 1 / 2 of the difference between the minimum sample S0 and the sample S1 , and the correction value Vadd added to the sample S2 is 1 / 2 of the difference between the sample S2 and the maximum sample S3 .

[0073] In addition, the even harmonic adding unit 171 subtracts the correction value Vsub from the sample S4, and the correction value Vsub is obtained by multiplying the difference between the maximum sample S3 and the sample S4 by a coefficient. The sample S4 is corrected to the sample S4' by subtracting the correction value Vsub. In addition, the even harmonic adding unit 171 adds the correction value Vadd to the sample S5, and the correction value Vadd is obtained by multiplying the difference between the sample S5 and the minimum sample S6 by a coefficient. The sample S5 is corrected to the sample S5' by adding the correction value Vadd.

[0074] The correction value Vsub subtracted from the sample S4 is 1 / 2 of the difference between the maximum sample S3 and the sample S4, and the correction value Vadd added to the sample S5 is 1 / 2 of the difference between the sample S5 and the minimum sample S6.

[0075] The odd harmonic adding unit 172 subtracts a correction value Vsub from the minimum sample S0, the correction value Vsub being obtained by multiplying the difference between the sample S99 and the minimum sample S0 by a coefficient. The minimum sample S0 is corrected to the minimum sample S0' by subtracting the correction value Vsub. If the sampling interval between the minimum sample S0 and the maximum sample adjacent to the front side of the minimum sample S0 is 3fs, the correction value Vsub subtracted from the minimum sample S0 is 1 / 2 of the difference between the minimum sample S0 and the sample S99.

[0076] Furthermore, the odd harmonic adding unit 172 adds a correction value Vadd to the maximum sample S3, which is obtained by multiplying the difference between the sample S2 and the maximum sample S3 by a coefficient. The maximum sample S3 is corrected to the maximum sample S3' by adding the correction value Vadd. The correction value Vadd added to the maximum sample S3 is 1 / 2 of the difference between the sample S2 and the maximum sample S3.

[0077] In addition, the odd harmonic adding unit 172 subtracts the correction value Vsub from the minimum sample S6, and the correction value Vsub is obtained by multiplying the difference between the sample S5 and the minimum sample S6 by a coefficient. The minimum sample S6 is corrected to the minimum sample S6' by subtracting the correction value Vsub. The correction value Vsub subtracted from the minimum sample S6 is 1 / 2 of the difference between the sample S5 and the minimum sample S6.

[0078] The harmonic component adding unit 17 corrects the waveform shown by the single-dot chain line to the waveform shown by the solid line by adding the correction value Vadd or subtracting the correction value Vsub from the sample of the correction object of the digital sound signal as described above. The even harmonic adding unit 171 and the odd harmonic adding unit 172 also correct the digital sound signal in the same manner after sampling S7. Thus, harmonic components including even harmonics and odd harmonics are added to the digital sound signal.

[0079] like Figure 4 or Figure 5 As shown, the digital sound processing device 100 can add higher harmonic components including both even harmonics and odd harmonics to the digital sound signal by correcting the waveform of both the waveform portion where the sampling value rises and the waveform portion where the sampling value falls.

[0080] The even harmonic adding unit 171 may be configured such that, even if the sampling interval between the minimum sampling and the maximum sampling is any one of 3fs to 8fs, only the sampling before and after the minimum sampling and the sampling before and after the maximum sampling are used as the samples to be corrected.

[0081] The even harmonic adding unit 171 may also add two samples before and two samples after the minimum sample, and two samples before and two samples after the maximum sample to the samples of the correction object when the sampling interval between the minimum sample and the maximum sample is, for example, 6fs to 8fs.

[0082] use Figure 6 , the operation of the even harmonic adding unit 171 is described in the case where a total of four samples, namely, two samples before and after a minimum sample and two samples before and after a maximum sample, are samples to be corrected. Figure 6 Equivalent to Figure 4 The additive method for even harmonics shown is extended to the additive method for four samples. Figure 6 In FIG. 1 , only the correction values ​​and the samples after correction are shown for the samples ranging from the minimum sample S0 to the next minimum sample S12.

[0083] The even harmonic adding unit 171 adds a correction value Vadd obtained by multiplying the difference between the minimum sample S0 and the sample S1 by a coefficient to the sample S1, and adds a correction value Vadd obtained by multiplying the difference between the sample S1 and the sample S2 by a coefficient to the sample S2. The samples S1 and S2 are corrected to samples S1' and S2' by adding the correction value Vadd. In addition, the even harmonic adding unit 171 subtracts a correction value Vsub obtained by multiplying the difference between the sample S4 and the sample S5 by a coefficient from the sample S4, and subtracts a correction value Vsub obtained by multiplying the difference between the sample S5 and the maximum sample S6 by a coefficient from the sample S5. The samples S4 and S5 are corrected to samples S4' and S5' by subtracting the correction value Vsub.

[0084] In addition, the even harmonic adding section 171 adds the correction value Vadd obtained by multiplying the difference between the maximum sample S6 and the sample S7 by a coefficient to the sample S7, and adds the correction value Vadd obtained by multiplying the difference between the sample S7 and the sample S8 by a coefficient to the sample S8. The samples S7 and S8 are corrected to samples S7' and S8' by adding the correction value Vadd. In addition, the even harmonic adding section 171 subtracts the correction value Vsub obtained by multiplying the difference between the sample S10 and the sample S11 by a coefficient from the sample S10, and subtracts the correction value Vsub obtained by multiplying the difference between the sample S11 and the extremely small sample S12 by a coefficient from the sample S11. The samples S10 and S11 are corrected to samples S10' and S11' by subtracting the correction value Vsub.

[0085] The odd harmonic adding unit 172 subtracts the correction value Vsub obtained by multiplying the difference between sample S99 and the minimum sample S0 by a coefficient from the minimum sample S0. The minimum sample S0 is corrected to the minimum sample S0' by subtracting the correction value Vsub. The odd harmonic adding unit 172 adds the correction value Vadd obtained by multiplying the difference between sample S5 and the maximum sample S6 by a coefficient to the maximum sample S6. The maximum sample S6 is corrected to the maximum sample S6' by adding the correction value Vadd. The odd harmonic adding unit 172 subtracts the correction value Vsub obtained by multiplying the difference between sample S11 and the minimum sample S12 by a coefficient from the minimum sample S12. The minimum sample S12 is corrected to the minimum sample S12' by subtracting the correction value Vsub.

[0086] The harmonic component adding unit 17 corrects the waveform shown by the dashed line to the waveform shown by the solid line by adding the correction value Vadd or subtracting the correction value Vsub from the correction target sample of the digital audio signal as described above.

[0087] like Figure 7As shown, the harmonic component adding unit 17 may correct the digital audio signal by adding the correction value Vadd or subtracting the correction value Vsub from the sample to be corrected. Figure 7 Equivalent to Figure 5 The additive method for even harmonics shown is expanded to an additive method for four samples.

[0088] The even harmonic adding section 171 subtracts the correction value Vsub obtained by multiplying the difference between the extremely small sample S0 and the sample S1 by a coefficient from the sample S1, and subtracts the correction value Vsub obtained by multiplying the difference between the sample S1 and the sample S2 by a coefficient from the sample S2. The samples S1 and S2 are respectively corrected to samples S1' and S2' by subtracting the correction value Vsub. In addition, the even harmonic adding section 171 adds the correction value Vadd obtained by multiplying the difference between the sample S4 and the sample S5 by a coefficient to the sample S4, and adds the correction value Vadd obtained by multiplying the difference between the sample S5 and the extremely large sample S6 by a coefficient to the sample S5. The samples S4 and S5 are respectively corrected to samples S4' and S5' by adding the correction value Vadd.

[0089] In addition, the even harmonic adding section 171 subtracts the correction value Vsub obtained by multiplying the difference between the maximum sample S6 and the sample S7 by a coefficient from the sample S7, and subtracts the correction value Vsub obtained by multiplying the difference between the sample S7 and the sample S8 by a coefficient from the sample S8. In addition, the even harmonic adding section 171 adds the correction value Vadd obtained by multiplying the difference between the sample S10 and the sample S11 by a coefficient to the sample S10, and adds the correction value Vadd obtained by multiplying the difference between the sample S11 and the minimum sample S12 by a coefficient to the sample S11. The samples S7 and S8 are corrected to the samples S7' and S8' respectively by subtracting the correction value Vsub. The samples S10 and S11 are corrected to the samples S10' and S11' respectively by adding the correction value Vadd.

[0090] The odd harmonic adding unit 172 subtracts the correction value Vsub obtained by multiplying the difference between sample S99 and the minimum sample S0 by a coefficient from the minimum sample S0. The minimum sample S0 is corrected to the minimum sample S0' by subtracting the correction value Vsub. The odd harmonic adding unit 172 adds the correction value Vadd obtained by multiplying the difference between sample S5 and the maximum sample S6 by a coefficient to the maximum sample S6. The maximum sample S6 is corrected to the maximum sample S6' by adding the correction value Vadd. The odd harmonic adding unit 172 subtracts the correction value Vsub obtained by multiplying the difference between sample S11 and the minimum sample S12 by a coefficient from the minimum sample S12. The minimum sample S12 is corrected to the minimum sample S12' by subtracting the correction value Vsub.

[0091] The harmonic component adding unit 17 corrects the waveform shown by the dashed line to the waveform shown by the solid line by adding the correction value Vadd or subtracting the correction value Vsub from the correction target sample of the digital audio signal as described above.

[0092] The even harmonic adding unit 171 may also use three or more samples before and after the minimum sample as samples for correction, and add or subtract the correction value Vadd or Vsub. The number of samples for correction is a matter of design. Alternatively, the longer the sampling interval is, the more samples for correction are increased.

[0093] However, when the sampling interval is 2fs, the intermediate samples between the minimum sample and the maximum sample become the samples to which the correction value Vadd is added and the samples to which the correction value Vsub is subtracted. Therefore, when the sampling interval is 2fs, the even harmonic adding section 171 is preferably as follows: Figure 8 or Fig. 9 The digital sound signal is corrected as shown.

[0094] Figure 8 Corresponds to Figure 4 The additional method for even harmonics is shown in Figure 8 In the example, the even harmonic adding unit 171 subtracts the correction value Vsub obtained by multiplying the difference between the sample S1 and the maximum sample S2 by a coefficient from the sample S1. The sample S1 is corrected to the sample S1' by subtracting the correction value Vsub. In addition, the even harmonic adding unit 171 adds the correction value Vadd obtained by multiplying the difference between the maximum sample S2 and the sample S3 by a coefficient to the sample S3. The sample S3 is corrected to the sample S3' by adding the correction value Vadd.

[0095] The odd harmonic adding unit 172 subtracts the correction value Vsub obtained by multiplying the difference between the sample S99 and the minimum sample S0 by a coefficient from the minimum sample S0. The minimum sample S0 is corrected to the minimum sample S0' by subtracting the correction value Vsub. The odd harmonic adding unit 172 adds the correction value Vadd obtained by multiplying the difference between the sample S1 and the maximum sample S2 by a coefficient to the maximum sample S2. The maximum sample S2 is corrected to the maximum sample S2' by adding the correction value Vadd. The odd harmonic adding unit 172 subtracts the correction value Vsub obtained by multiplying the difference between the sample S3 and the minimum sample S4 by a coefficient from the minimum sample S4. The minimum sample S4 is corrected to the minimum sample S4' by subtracting the correction value Vsub.

[0096] Fig. 9 Corresponds to Figure 5 The additional method for even harmonics is shown in Fig. 9In the example, the even harmonic adding unit 171 adds a correction value Vadd obtained by multiplying the difference between the sample S1 and the maximum sample S2 by a coefficient to the sample S1. The sample S1 is corrected to the sample S1' by adding the correction value Vadd. In addition, the even harmonic adding unit 171 subtracts a correction value Vsub obtained by multiplying the difference between the maximum sample S2 and the sample S3 by a coefficient from the sample S3. The sample S3 is corrected to the sample S3' by subtracting the correction value Vsub. The operation of the odd harmonic adding unit 172 is the same as that of the Figure 8 same.

[0097] The operation of the digital sound processing device 100 for processing a digital sound signal with a sampling interval of 3fs or more is summarized as follows. Assume that the samples constituting the digital sound signal rise from the first minimum sample to the first maximum sample, and then fall from the first maximum sample to the second minimum sample. Figure 4 and Figure 5 In , the minimum sample S0 is the first minimum sample, the maximum sample S3 is the first maximum sample, and the minimum sample S6 is the second minimum sample. Figure 6 and Figure 7 In the figure, the minimum sample S0 is the first minimum sample, the maximum sample S6 is the first maximum sample, and the minimum sample S12 is the second minimum sample.

[0098] The even harmonic adding section 171 performs any one of a first even harmonic adding process and a second even harmonic adding process.

[0099] The first even harmonic addition process is as follows. The even harmonic addition unit 171 adds the first correction value to the first adjacent sample, the first adjacent sample is the next sample after the first minimum sample, and the first correction value is obtained by multiplying the first difference between the first minimum sample and the first adjacent sample by the first even harmonic coefficient. The coefficient selection unit 15 selects the first even harmonic coefficient according to the first sampling interval between the first minimum sample and the first maximum sample. The even harmonic addition unit 171 subtracts the second correction value from the second adjacent sample that is one sample before the first maximum sample, and the second correction value is obtained by multiplying the second difference between the second adjacent sample and the first maximum sample by the first even harmonic coefficient.

[0100] The even harmonic adding unit 171 adds the third correction value to the third adjacent sample, the third adjacent sample is the next sample after the first maximum sample, and the third correction value is obtained by multiplying the third difference between the first maximum sample and the third adjacent sample by the second even harmonic coefficient. The coefficient selecting unit 15 selects the second even harmonic coefficient according to the second sampling interval between the first maximum sample and the second minimum sample. The second even harmonic coefficient can be the same coefficient as the first even harmonic coefficient, or it can be a different coefficient. The even harmonic adding unit 171 subtracts the fourth correction value from the fourth adjacent sample that is one sample before the second minimum sample, and the fourth correction value is obtained by multiplying the fourth difference between the fourth adjacent sample and the second minimum sample by the second even harmonic coefficient.

[0101] The second even harmonic addition process is as follows: The even harmonic addition section 171 subtracts the first correction value from the first adjacent sample and adds the second correction value to the second adjacent sample. The even harmonic addition section 171 subtracts the third correction value from the third adjacent sample and adds the fourth correction value to the fourth adjacent sample.

[0102] The odd harmonic adding unit 172 performs the following odd harmonic adding process. The odd harmonic adding unit 172 subtracts a fifth correction value from the first minimum sample, the fifth correction value being a fifth difference between the first minimum sample and the fifth adjacent sample one sample before the first minimum sample multiplied by the first odd harmonic coefficient. The coefficient selecting unit 15 selects the first odd harmonic coefficient based on the third sampling interval between the first minimum sample and the second maximum sample immediately before the first minimum sample.

[0103] The odd harmonic adding unit 172 adds a sixth correction value obtained by multiplying the second difference by the second odd harmonic coefficient to the first maximum sampling. The second odd harmonic coefficient is selected by the coefficient selecting unit 15 according to the first sampling interval. The second odd harmonic coefficient is the same as the first even harmonic coefficient. The odd harmonic adding unit 172 subtracts a seventh correction value obtained by multiplying the fourth difference by the third odd harmonic coefficient from the second minimum sampling. The third odd harmonic coefficient is selected by the coefficient selecting unit 15 according to the second sampling interval. The third odd harmonic coefficient is the same as the second even harmonic coefficient.

[0104] As described above, the harmonic component adding section 17 including the even harmonic adding section 171 and the odd harmonic adding section 172 adds harmonic components including even harmonics and odd harmonics to the input digital audio signal and outputs the resultant signal.

[0105] use Fig.10 The flowchart shown in FIG. 1 illustrates a digital sound processing method as a process executed by the digital sound processing device 100. Fig.10In the present invention, when a digital sound signal is input to the digital sound processing device 100 and the processing starts, the maximum sample detection unit 11 and the minimum sample detection unit 12 detect the maximum sample and the minimum sample in step S01. The waveform inclination determination unit 13 determines in step S02 whether the samples constituting the digital sound signal are the waveform part where the sample value rises from the minimum sample to the maximum sample, or the waveform part where the sample value falls from the maximum sample to the minimum sample.

[0106] In parallel with step S02 , the counter 14 counts the sampling intervals between the minimum sample and the maximum sample adjacent in the time direction in step S03 . In step S04 , the coefficient selection unit 15 selects the even harmonic coefficients and the odd harmonic coefficients according to the sampling intervals counted by the counter 14 .

[0107] In step S05, the even harmonic adding unit 171 and the odd harmonic adding unit 172 calculate the correction values ​​Vadd and Vsub as the correction values ​​for the even harmonics, and calculate the correction values ​​Vadd and Vsub as the correction values ​​for the odd harmonics. In step S06, the even harmonic adding unit 171 and the odd harmonic adding unit 172 add and subtract the correction values ​​for the even harmonics and the correction values ​​for the odd harmonics to the digital sound signal to correct the waveform of the digital sound signal.

[0108] In step S07, the digital sound processing device 100 determines whether the input of the digital sound signal has been completed. If the input of the digital sound signal has not been completed (No), the digital sound processing device 100 repeats the processing of steps S01 to S07. If the input of the digital sound signal has been completed (Yes), the digital sound processing device 100 ends the processing.

[0109] However, at a sampling interval of 3fs Figure 4 In the first even harmonic addition process shown, the correction value Vadd is added to the sample S1 of the adjacent samples S1 and S2, and the correction value Vsub is subtracted from the sample S2. Therefore, depending on the magnitude relationship between the correction value Vadd and the correction value Vsub, a reversal phenomenon may occur in which the magnitude relationship between the samples S1' and S2' is reversed. Similarly, in Figure 5 In the second even harmonic addition process shown, a reversal phenomenon may occur in which the magnitude relationship between the sample S4 ′ and the sample S5 ′ is reversed.

[0110] use Fig.11A as well as Fig. 11B , taking the first even harmonic additional processing as an example, the situation where the magnitude relationship between sampling S1' and sampling S2' is reversed is explained. Fig.11AIn the example, the difference between the minimum sample S0 and the sample S1 is set to Δ01, the difference between the sample S1 and the sample S2 is set to Δ12, and the difference between the sample S2 and the maximum sample S3 is set to Δ23. The difference Δ23 is significantly larger than the differences Δ01 and Δ12.

[0111] In this case, if Fig. 11B As shown, the correction value Vsub subtracted from the sample S2 is significantly larger than the correction value Vadd added to the sample S1. Fig. 11B This indicates the case where the coefficient is set to 1 / 2. Therefore, the waveform in which the sampling value rises from sample S1 to sample S2 becomes the waveform in which the sampling value falls from sample S1' to sample S2'. Such an inversion phenomenon of the sampling values ​​of sample S1' and sample S2' will damage the original waveform before correction, so it is preferable to avoid the occurrence of the inversion phenomenon.

[0112] When the sampling interval is longer than 4fs, the inversion phenomenon may also occur. Fig. 12A In the example, the difference between the minimum sample S0 and the sample S1 is set to Δ01, the difference between the sample S1 and the sample S2 is set to Δ12, the difference between the sample S2 and the sample S3 is set to Δ23, and the difference between the sample S3 and the maximum sample S4 is set to Δ34. The difference Δ01 is significantly larger than the difference Δ12, and the difference Δ34 is significantly larger than the difference Δ23.

[0113] In such a case, Fig. 12B As shown, the correction value Vadd added to the sample S1 is larger than the difference Δ12, and the sampling value of the sample S1' is larger than the sampling value of the sample S2. In addition, the correction value Vsub subtracted from the sample S3 is larger than the difference Δ23, and the sampling value of the sample S3' is smaller than the sampling value of the sample S2. Fig. 12B This indicates the case where the coefficient is set to 1 / 2.

[0114] When the sampling interval is 4fs or more, it is preferable to avoid the inversion phenomenon that the sampling value of the correction sample to which the correction value Vadd is added becomes larger than the sampling value of the next sample. In addition, it is preferable to avoid the inversion phenomenon that the sampling value of the correction sample to which the correction value Vsub is subtracted becomes smaller than the sampling value of the previous sample.

[0115] Therefore, when the sampling interval is 3fs and the first adjacent sample and the second adjacent sample are adjacent, the even harmonic adding unit 171 can perform the first even harmonic adding process as follows. The even harmonic adding unit 171 limits the first correction value and the second correction value so that the magnitude relationship of the sampling values ​​between the first correction sample and the second correction sample is not reversed, the first correction sample is obtained by adding the first correction value to the first adjacent sample, and the second correction sample is obtained by subtracting the second correction value from the second adjacent sample.

[0116] In addition, the even harmonic adding unit 171 may perform the second even harmonic adding process in the following manner: the even harmonic adding unit 171 limits the third correction value and the fourth correction value so that the magnitude relationship of the sampling values ​​between the third correction sample obtained by subtracting the third correction value from the third adjacent sample and the fourth correction sample obtained by adding the fourth correction value to the fourth adjacent sample is not reversed.

[0117] Specifically, in order to avoid the occurrence of the inversion phenomenon when the sampling interval is 3fs, the digital sound processing device 100 can include Fig.13 The processing shown is performed in the following manner Fig.10 Steps S05 and S06. Fig.13 This is the first example of processing to avoid the inversion phenomenon in the case of a waveform with a sampling interval of 3 fs, taking the first even harmonic addition processing as an example. The processing to avoid the occurrence of the inversion phenomenon in the second even harmonic addition processing is the same.

[0118] In step S501, the even harmonic adding unit 171 calculates the difference Δ01 between the minimum sample S0 and the sample S1, the difference Δ12 between the sample S1 and the sample S2, and the difference Δ23 between the sample S2 and the maximum sample S3. In step S502, the even harmonic adding unit 171 calculates the maximum correction values ​​Vaddmax and Vsubmax, and in step S503, calculates the correction values ​​Vadd and Vsub. The order of step S502 and step S503 can also be reversed. As a first example, the maximum correction values ​​Vaddmax and Vsubmax are set to 1 / 2 of the difference Δ12. The maximum correction values ​​Vaddmax and Vsubmax can also be set to values ​​less than 1 / 2 of the difference Δ12.

[0119] In step S601, the even harmonic adding unit 171 determines whether the correction value Vadd exceeds the maximum correction value Vaddmax. If the correction value Vadd does not exceed the maximum correction value Vaddmax (No), the even harmonic adding unit 171 selects the correction value Vadd in step S602, and the process is transferred to step S604. If the correction value Vadd exceeds the maximum correction value Vaddmax (Yes), the even harmonic adding unit 171 selects the maximum correction value Vaddmax in step S603, and the process is transferred to step S604.

[0120] In step S604, the even harmonic adding unit 171 determines whether the correction value Vsub exceeds the maximum correction value Vsubmax. If the correction value Vsub does not exceed the maximum correction value Vsubmax (No), the even harmonic adding unit 171 selects the correction value Vsub in step S605, and the process is transferred to step S607. If the correction value Vsub exceeds the maximum correction value Vsubmax (Yes), the even harmonic adding unit 171 selects the maximum correction value Vsubmax in step S606, and the process is transferred to step S607.

[0121] In step S607 , the even harmonic adding section 171 adds the correction value Vadd or the maximum correction value Vaddmax to the sample S1 , and subtracts the correction value Vsub or the maximum correction value Vsubmax from the sample S2 .

[0122] In this way, since the correction values ​​Vadd and Vsub do not exceed the median value of the sampling S1 and the sampling S2, the occurrence of the inversion phenomenon is avoided.

[0123] As a second example, the maximum correction values ​​Vaddmax and Vsubmax may be set to values ​​obtained by dividing the difference Δ12 by the ratio of the difference Δ01 to the difference Δ23. The maximum correction value Vaddmax is calculated by (Δ01×Δ12) / (Δ01+Δ23), and the maximum correction value Vsubmax is calculated by (Δ23×Δ12) / (Δ01+Δ23). In this way, the correction values ​​Vadd and Vsub do not exceed the value obtained by dividing the difference Δ12 by the ratio of the difference Δ01 to the difference Δ23, thereby avoiding the occurrence of the inversion phenomenon.

[0124] exist Fig.13 In the case where, for example, the correction value Vadd is small and the correction value Vsub is large, the sum of the correction value Vadd and the correction value Vsub may not exceed the difference Δ12. In this case, the inversion phenomenon does not actually occur. Regardless of whether the inversion phenomenon actually occurs, if the correction value Vadd exceeds the maximum correction value Vaddmax, the correction value Vadd is limited to the maximum correction value Vaddmax, and if the correction value Vsub exceeds the maximum correction value Vsubmax, the correction value Vsub is limited to the maximum correction value Vsubmax.

[0125] In order to avoid the occurrence of the inversion phenomenon when the sampling interval is 3fs, the digital sound processing device 100 may also replace Fig.13 The processing shown is to include Fig.14 The processing shown is performed in the following manner Fig.10 Steps S05 and S06. Fig.14This is a second example of processing to avoid the inversion phenomenon in the case of a waveform with a sampling interval of 3 fs, taking the first even harmonic addition processing as an example. The processing to avoid the occurrence of the inversion phenomenon in the second even harmonic addition processing is the same.

[0126] Fig.14 Steps S501 to S503 in Fig.13 The steps S501 to S503 are the same. Fig.14 In the example, the maximum correction values ​​Vaddmax and Vsubmax may be 1 / 2 of the difference Δ12, or may be a value obtained by dividing the difference Δ12 by the ratio of the difference Δ01 to the difference Δ23.

[0127] exist Fig.14 In step S611, the even harmonic adding unit 171 determines whether the added value of the correction value Vadd and the correction value Vsub exceeds the difference Δ12. The added value exceeding the difference Δ12 means that an inversion phenomenon occurs. If the added value does not exceed the difference Δ12 (no), the even harmonic adding unit 171 selects the correction value Vadd and Vsub in step S612. Then, in step S614, the even harmonic adding unit 171 adds the correction value Vadd to the sample S1 and subtracts the correction value Vsub from the sample S2.

[0128] If the added value exceeds the difference Δ12 (Yes) in step S611, the even harmonic adding section 171 selects the maximum correction value Vaddmax and Vsubmax in step S613. Then, in step S615, the even harmonic adding section 171 adds the maximum correction value Vaddmax to the sample S1 and subtracts the maximum correction value Vsubmax from the sample S2.

[0129] exist Fig.14 In the case where the sum of the correction value Vadd and the correction value Vsub exceeds the difference Δ12 and an inversion phenomenon actually occurs, the correction value Vadd is limited to the maximum correction value Vaddmax, and the correction value Vsub is limited to the maximum correction value Vsubmax.

[0130] Fig.15A Shown by Fig.13 or Fig.14 The processing shown is used to avoid the occurrence of the inversion phenomenon. Fig.15A The waveform before correction is shown Fig.11A The waveform is the same. Fig.15A This shows the case where the maximum correction values ​​Vaddmax and Vsubmax are set to 1 / 2 of the difference Δ12. In this case, the samples S1' and S2' have the same value, and the waveform is flat. Fig. 15BThis shows a case where the maximum correction values ​​Vaddmax and Vsubmax are set to values ​​smaller than 1 / 2 of the difference Δ12. In this case, since the sample S1' is a smaller value than the sample S2', the inclined waveform is maintained.

[0131] As described above, when the even harmonic adding unit 171 performs the first even harmonic adding process when the first sampling interval is 3fs and the first adjacent sample is adjacent to the second adjacent sample, the occurrence of the inversion phenomenon can be avoided in the following manner. The even harmonic adding unit 171 limits the first correction value and the second correction value so that the magnitude relationship of the sampling values ​​between the first correction sample and the second correction sample is not reversed. The first correction sample is obtained by adding the first correction value to the first adjacent sample, and the second correction sample is obtained by subtracting the second correction value from the second adjacent sample.

[0132] When the even harmonic adding unit 171 performs the second even harmonic adding process when the second sampling interval is 3fs and the third adjacent sample is adjacent to the fourth adjacent sample, the occurrence of the inversion phenomenon can be avoided as described below. The even harmonic adding unit 171 limits the third correction value and the fourth correction value so that the magnitude relationship of the sampling values ​​between the third correction sample and the fourth correction sample is not reversed. The third correction sample is obtained by subtracting the third correction value from the third adjacent sample, and the fourth correction sample is obtained by adding the fourth correction value to the fourth adjacent sample.

[0133] In order to avoid the occurrence of the inversion phenomenon when the sampling interval is greater than 4fs, the digital sound processing device 100 may include Fig.16 The processing shown is performed in the following manner Fig.10 Steps S05 and S06. Fig.16 The even harmonic adding unit 171 shows a case where the correction value Vadd is added or subtracted from the correction value Vsub only to two samples sandwiching a minimum sample and two samples sandwiching a maximum sample. Fig.16 Taking the first even-order harmonic addition process as an example, the process for avoiding the occurrence of the inversion phenomenon in the second even-order harmonic addition process is the same.

[0134] The maximum sample is set as Sn, the sample before the maximum sample Sn is set as S(n-1), and the two samples before it are set as S(n-2). The difference between the sample S(n-2) and the sample S(n-1) is set as Δ(n-2, n-1), and the difference between the sample S(n-1) and the maximum sample Sn is set as Δ(n-1, n).

[0135] exist Fig.16In step S511, the even harmonic adding unit 171 calculates the difference Δ01 between the minimum sample S0 and the sample S1, the difference Δ12 between the sample S1 and the sample S2, the difference Δ(n-2, n-1) between the sample S(n-2) and the sample S(n-1), and the difference Δ(n-1, n) between the sample S(n-1) and the maximum sample Sn.

[0136] In step S512, the even harmonic adding unit 171 sets the difference Δ12 as the maximum correction value Vaddmax, and sets the difference Δ(n-2, n-1) as the maximum correction value Vsubmax. In step S513, the even harmonic adding unit 171 calculates the correction values ​​Vadd and Vsub. The order of step S512 and step S513 may also be reversed. It is also possible to multiply the difference Δ12 by a value less than 1 and set the value less than the difference Δ12 as the maximum correction value Vaddmax, and multiply the difference Δ(n-2, n-1) by a value less than 1 and set the value less than the difference Δ(n-2, n-1) as the maximum correction value Vsubmax.

[0137] In step S621, the even harmonic adding unit 171 determines whether the correction value Vadd exceeds the maximum correction value Vaddmax. If the correction value Vadd does not exceed the maximum correction value Vaddmax (No), the even harmonic adding unit 171 selects the correction value Vadd in step S622, and the process is transferred to step S624. If the correction value Vadd exceeds the maximum correction value Vaddmax (Yes), the even harmonic adding unit 171 selects the maximum correction value Vaddmax in step S623, and the process is transferred to step S624.

[0138] In step S624, the even harmonic adding unit 171 determines whether the correction value Vsub exceeds the maximum correction value Vsubmax. If the correction value Vsub does not exceed the maximum correction value Vsubmax (No), the even harmonic adding unit 171 selects the correction value Vsub in step S625, and the process is transferred to step S627. If the correction value Vsub exceeds the maximum correction value Vsubmax (Yes), the even harmonic adding unit 171 selects the maximum correction value Vsubmax in step S626, and the process is transferred to step S627.

[0139] In step S627 , the even harmonic adding section 171 adds the correction value Vadd or the maximum correction value Vaddmax to the sample S1 , and subtracts the correction value Vsub or the maximum correction value Vsubmax from the sample S2 .

[0140] pass Fig.16 The processing shown, such as Fig.17A or Fig. 17BAs shown, the occurrence of the inversion phenomenon that the sampling value of the sample S1' is larger than the sampling value of the sample S2 is avoided. In addition, the occurrence of the inversion phenomenon that the sampling value of the sample S3' is smaller than the sampling value of the sample S2 is avoided. Fig.17A and Fig. 17B Indicates the case where the sampling interval is 4fs.

[0141] Fig.17A The case where the difference Δ12 is set as the maximum correction value Vaddmax and the difference Δ23 is set as the maximum correction value Vsubmax is shown. In this case, the samples S1 ′, S2 ′, and S3 ′ have the same value, and the waveform becomes flat. Fig. 17B The maximum correction value Vaddmax is set to a value smaller than the difference Δ12 and the maximum correction value Vsubmax is set to a value smaller than the difference Δ23. In this case, since the sample S1' is smaller than the sample S2 and the sample S3' is larger than the sample S2, the inclined waveform is maintained.

[0142] As described above, when the first even harmonic addition process is performed when the first sampling interval is 4fs or more, the even harmonic addition unit 171 can avoid the occurrence of the inversion phenomenon as follows. The even harmonic addition unit 171 limits the first correction value so that the sampling value of the first correction sampling is not greater than the sampling value of the next sampling after the first adjacent sampling. In addition, the even harmonic addition unit 171 limits the second correction value so that the sampling value of the second correction sampling is not less than the sampling value of the sampling one sampling before the second adjacent sampling.

[0143] When the even harmonic adding unit 171 performs the second even harmonic adding process when the second sampling interval is 4fs or more, the occurrence of the inversion phenomenon can be avoided as described below. The even harmonic adding unit 171 limits the third correction value so that the sampling value of the third correction sampling is not less than the sampling value of the next sampling after the third adjacent sampling. In addition, the even harmonic adding unit 171 limits the fourth correction value so that the sampling value of the fourth correction sampling is not greater than the sampling value of the sampling one sampling before the fourth adjacent sampling.

[0144] like Figure 6 or Figure 7 As shown, when the even harmonic adding unit 171 adds the correction value Vadd or subtracts the correction value Vsub to four samples each of two minimum samples and four samples each of two maximum samples, the even harmonic adding unit 171 can also avoid the occurrence of the inversion phenomenon as follows.

[0145] exist Figure 6In the example, the even harmonic adding unit 171 limits the first correction value so that the sampling value of the correction sample S1' is not greater than the sampling value of the next correction sample S2'. The even harmonic adding unit 171 limits the second correction value so that the sampling value of the correction sample S5' is not less than the sampling value of the correction sample S4' of the previous sampling. Figure 7 The even harmonic adding unit 171 limits the third correction value so that the sampling value of the correction sample S7' is not less than the sampling value of the next correction sample S8'. The even harmonic adding unit 171 limits the fourth correction value so that the sampling value of the correction sample S11' is not greater than the sampling value of the previous correction sample S10'.

[0146] When the number of correction target samples further increases, similarly, the even harmonic adding unit 171 only needs to limit the correction value so as not to be larger or smaller than the sample value of the correction sample adjacent to each correction target sample.

[0147] Figure 1 The digital sound processing device 100 shown can be realized by a central processing unit (CPU) of a microcomputer executing a digital sound processing program. Fig.18 In the embodiment, the CPU 50, the main memory 55 and the storage medium 60 are connected by a bus. The storage medium 60 is any non-temporary storage medium such as a hard disk drive, an optical disk, a semiconductor memory, etc. The digital sound processing program is stored in the storage medium 60. The digital sound processing program can also be sent from an external server via a communication line such as the Internet and stored in the storage medium 60.

[0148] The CPU 50 loads the digital sound processing program stored in the storage medium 60 into the main memory 55. The CPU 50 executes the digital sound processing program by executing each instruction described in the digital sound processing program loaded into the main memory 55. Fig.10 The CPU 50 executes any one of the first even harmonic adding step corresponding to the above-described first even harmonic adding process and the second even harmonic adding step corresponding to the second even harmonic adding process, and the odd harmonic adding step corresponding to the above-described odd harmonic adding process.

[0149] When the sampling interval of CPU 50 is 3fs, it is preferred to set Fig.13 or Fig.14 The CPU 50 preferably sets the same as the processing shown in the figure, which is equivalent to the step of limiting the first correction value and the second correction value, or the step of limiting the third correction value and the fourth correction value, to avoid the occurrence of the inversion phenomenon. When the sampling interval is greater than 4fs, the CPU 50 preferably sets the same as the processing shown in the figure. Fig.16 The processing shown corresponds to a step of limiting the first correction value and the second correction value, or a step of limiting the third correction value and the fourth correction value.

[0150] The present invention is not limited to the above-described embodiment, and various modifications can be made without departing from the gist of the present invention.

[0151] This application claims priority based on Japanese Patent Application No. 2019-146149 filed with the Japan Patent Office on August 8, 2019, the disclosure of which is incorporated herein by reference in its entirety.

Claims

1. A digital sound processing device, comprising: A maximum sample detection unit that calculates a maximum value among samples constituting an input digital sound signal and detects a maximum sample having a maximum value; A minimum sample detection unit that calculates a minimum value among samples constituting the digital sound signal and detects a minimum sample having a minimum value; A waveform tilt determination unit that determines whether the samples constituting the digital sound signal are waveform portions where the sampling values ​​rise from a minimum sampling value to a maximum sampling value, or waveform portions where the sampling values ​​fall from a maximum sampling value to a minimum sampling value; A counter, for counting the sampling intervals between the minimum sampling and the maximum sampling adjacent to each other in the time direction; A coefficient selection unit, which selects coefficients for even harmonics and coefficients for odd harmonics according to the sampling interval counted by the counter; as well as a higher harmonic component adding unit, comprising an even harmonic adding unit for adding even harmonics to the digital sound signal and an odd harmonic adding unit for adding odd harmonics, adding higher harmonic components consisting of even harmonics and odd harmonics to the digital sound signal and outputting the resultant signal; wherein, when the samples constituting the digital sound signal rise from a first minimum sample to a first maximum sample and fall from the first maximum sample to a second minimum sample, the even harmonic adding section performs any one of a first even harmonic adding process and a second even harmonic adding process, In the first even harmonic addition process, even harmonics are added to the digital sound signal by: adding a first correction value to a first adjacent sample, the first adjacent sample being a next sample after the first minimum sample, the first correction value being obtained by multiplying a first difference between the first minimum sample and the first adjacent sample by a first even harmonic coefficient, the first even harmonic coefficient being selected by the coefficient selection unit according to a first sampling interval between the first minimum sample and the first maximum sample; Subtracting a second correction value from a second adjacent sample, the second adjacent sample being a sample preceding the first maximum sample, the second correction value being obtained by multiplying a second difference between the second adjacent sample and the first maximum sample by the first even harmonic coefficient; adding a third correction value to a third adjacent sample, the third adjacent sample being a next sample after the first maximum sample, the third correction value being obtained by multiplying a third difference between the first maximum sample and the third adjacent sample by a second even harmonic coefficient, the second even harmonic coefficient being selected by the coefficient selection unit according to a second sampling interval between the first maximum sample and the second minimum sample; and A fourth correction value is subtracted from a fourth adjacent sample, the fourth adjacent sample being a sample preceding the second minimum sample, the fourth correction value being obtained by multiplying a fourth difference between the fourth adjacent sample and the second minimum sample by the second even harmonic coefficient, and in the second even harmonic addition processing, even harmonics are added to the digital sound signal by: subtracting the first correction value from the first adjacent samples; adding the second correction value to the second adjacent samples; subtracting the third correction value from the third adjacent sample; and adding the fourth correction value to the fourth adjacent sample, Furthermore, when the samples constituting the digital sound signal rise from the first minimum sample to the first maximum sample and fall from the first maximum sample to the second minimum sample, the odd harmonic adding unit performs an odd harmonic adding process of adding odd harmonics to the digital sound signal by: subtracting a fifth correction value from the first minimum sample, the fifth correction value being obtained by multiplying a fifth difference between the first minimum sample and a fifth adjacent sample by a first odd harmonic coefficient, the fifth adjacent sample being a sample preceding the first minimum sample, the first odd harmonic coefficient being selected by the coefficient selection unit according to a third sampling interval, the third sampling interval being an interval between the first minimum sample and a second maximum sample immediately preceding the first minimum sample; adding a sixth correction value to the first maximum sample, the sixth correction value being obtained by multiplying the second difference by a second odd harmonic coefficient, the second odd harmonic coefficient being selected by the coefficient selection unit according to the first sampling interval; and A seventh correction value is subtracted from the second minimum sample, wherein the seventh correction value is obtained by multiplying the fourth difference by a third odd harmonic coefficient, wherein the third odd harmonic coefficient is selected by the coefficient selection unit according to the second sampling interval.

2. The digital sound processing device according to claim 1, wherein: When the first sampling interval is 3 and the first adjacent samples are adjacent to the second adjacent samples and the first even harmonic addition process is performed, The even harmonic adding unit limits the first correction value and the second correction value so that the magnitude relationship between the sampling values ​​of the first correction sample and the second correction sample is not reversed, the first correction sample is obtained by adding the first correction value to the first adjacent sample, and the second correction sample is obtained by subtracting the second correction value from the second adjacent sample, When the second sampling interval is 3 and the third adjacent sampling is adjacent to the fourth adjacent sampling and the second even harmonic addition process is performed, The even harmonic adding unit limits the third correction value and the fourth correction value so that the size relationship between the sampling values ​​of the third correction sample and the fourth correction sample is not reversed. The third correction sample is obtained by subtracting the third correction value from the third adjacent sample, and the fourth correction sample is obtained by adding the fourth correction value to the fourth adjacent sample.

3. The digital sound processing device according to claim 1 or 2, wherein: When the first sampling interval is greater than 4 and the first even harmonic addition processing is performed, the even harmonic addition unit limits the first correction value so that the sampling value of the first correction sample obtained by adding the first correction value to the first adjacent sample is not greater than the sampling value of the next sample after the first adjacent sample, and limits the second correction value so that the sampling value of the second correction sample obtained by subtracting the second correction value from the second adjacent sample is not less than the sampling value of the previous sample of the second adjacent sample. When the second sampling interval is greater than 4 and the second even harmonic addition processing is performed, the even harmonic addition unit limits the third correction value so that the sampling value of the third correction sample obtained by subtracting the third correction value from the third adjacent sample is not less than the sampling value of the next sample after the third adjacent sample, and limits the fourth correction value so that the sampling value of the fourth correction sample obtained by adding the fourth correction value to the fourth adjacent sample is not greater than the sampling value of the previous sample of the fourth adjacent sample.

4. A digital sound processing method, comprising: Calculating a maximum value among samples constituting an input digital sound signal and detecting a maximum sample having a maximum value; Calculating a minimum value among samples constituting the digital sound signal and detecting a minimum sample having a minimum value; Determine whether the samples constituting the digital sound signal are a waveform portion in which the sampling value increases from a minimum sampling value to a maximum sampling value, or a waveform portion in which the sampling value decreases from a maximum sampling value to a minimum sampling value; as well as Counting the sampling intervals between the minimum sampling and the maximum sampling adjacent to each other in the time direction; Selecting coefficients for even harmonics and odd harmonics according to the sampling interval obtained by counting; and adding a higher harmonic component consisting of even harmonics and odd harmonics to the digital sound signal and outputting the resultant signal; When the samples constituting the digital sound signal rise from a first minimum sample to a first maximum sample and fall from the first maximum sample to a second minimum sample, performing any one of a first even harmonic addition process and a second even harmonic addition process, In the first even harmonic addition process, even harmonics are added to the digital sound signal by: Adding a first correction value to a first adjacent sample, where the first adjacent sample is a next sample after the first minimum sample, and the first correction value is obtained by multiplying a first difference between the first minimum sample and the first adjacent sample by a first even harmonic coefficient, where the first even harmonic coefficient is selected according to a first sampling interval between the first minimum sample and the first maximum sample; Subtracting a second correction value from a second adjacent sample, the second adjacent sample being a sample preceding the first maximum sample, the second correction value being obtained by multiplying a second difference between the second adjacent sample and the first maximum sample by the first even harmonic coefficient; Adding a third correction value to a third adjacent sample, the third adjacent sample being a next sample after the first maximum sample, the third correction value being obtained by multiplying a third difference between the first maximum sample and the third adjacent sample by a second even harmonic coefficient, the second even harmonic coefficient being obtained by selecting according to a second sampling interval between the first maximum sample and the second minimum sample; as well as A fourth correction value is subtracted from a fourth adjacent sample, the fourth adjacent sample being a sample preceding the second minimum sample, the fourth correction value being obtained by multiplying a fourth difference between the fourth adjacent sample and the second minimum sample by the second even harmonic coefficient, and in the second even harmonic addition processing, even harmonics are added to the digital sound signal by: subtracting the first correction value from the first adjacent samples; adding the second correction value to the second adjacent samples; subtracting the third correction value from the third adjacent sample; as well as adding the fourth correction value to the fourth adjacent sample, Furthermore, when the samples constituting the digital sound signal rise from the first minimum sample to the first maximum sample and fall from the first maximum sample to the second minimum sample, an odd harmonic adding process of adding an odd harmonic to the digital sound signal is performed by: subtracting a fifth correction value from the first minimum sample, the fifth correction value being obtained by multiplying a fifth difference between the first minimum sample and a fifth adjacent sample by a first odd harmonic coefficient, the fifth adjacent sample being a sample preceding the first minimum sample, the first odd harmonic coefficient being selected according to a third sampling interval, the third sampling interval being an interval between the first minimum sample and a second maximum sample immediately preceding the first minimum sample; Adding a sixth correction value to the first maximum sampling, wherein the sixth correction value is obtained by multiplying the second difference by a second odd harmonic coefficient, wherein the second odd harmonic coefficient is selected according to the first sampling interval; as well as A seventh correction value is subtracted from the second minimum sample, wherein the seventh correction value is obtained by multiplying the fourth difference by a third odd harmonic coefficient, wherein the third odd harmonic coefficient is selected according to the second sampling interval.

5. A computer-readable storage medium storing a digital sound processing program, wherein the digital sound processing program, when executed, causes a computer to perform the following steps: Calculating a maximum value among samples constituting an input digital sound signal and detecting a maximum sample having a maximum value; Calculating a minimum value among samples constituting the digital sound signal and detecting a minimum sample having a minimum value; Determine whether the samples constituting the digital sound signal are a waveform portion in which the sampling value increases from a minimum sampling value to a maximum sampling value, or a waveform portion in which the sampling value decreases from a maximum sampling value to a minimum sampling value; as well as Counting the sampling intervals between the minimum sampling and the maximum sampling adjacent to each other in the time direction; Selecting coefficients for even harmonics and odd harmonics according to the sampling interval obtained by counting; and adding a higher harmonic component consisting of even harmonics and odd harmonics to the digital sound signal and outputting the resultant signal; When the samples constituting the digital sound signal rise from a first minimum sample to a first maximum sample and fall from the first maximum sample to a second minimum sample, performing any one of a first even harmonic adding step and a second even harmonic adding step, The first even harmonic adding step adds even harmonics to the digital sound signal, comprising the following steps: Adding a first correction value to a first adjacent sample, where the first adjacent sample is a next sample after the first minimum sample, and the first correction value is obtained by multiplying a first difference between the first minimum sample and the first adjacent sample by a first even harmonic coefficient, where the first even harmonic coefficient is selected according to a first sampling interval between the first minimum sample and the first maximum sample; Subtracting a second correction value from a second adjacent sample, the second adjacent sample being a sample preceding the first maximum sample, the second correction value being obtained by multiplying a second difference between the second adjacent sample and the first maximum sample by the first even harmonic coefficient; adding a third correction value to a third adjacent sample, the third adjacent sample being a next sample after the first maximum sample, the third correction value being obtained by multiplying a third difference between the first maximum sample and the third adjacent sample by a second even harmonic coefficient, the second even harmonic coefficient being selected according to a second sampling interval between the first maximum sample and the second minimum sample; and Subtracting a fourth correction value from a fourth adjacent sample, the fourth adjacent sample being a sample preceding the second minimum sample, the fourth correction value being obtained by multiplying a fourth difference between the fourth adjacent sample and the second minimum sample by the second even harmonic coefficient, the second even harmonic addition processing adding even harmonics to the digital sound signal, comprising the following steps: subtracting the first correction value from the first adjacent samples; adding the second correction value to the second adjacent samples; subtracting the third correction value from the third adjacent sample; and adding the fourth correction value to the fourth adjacent sample, Furthermore, when the samples constituting the digital sound signal rise from the first minimum sample to the first maximum sample and fall from the first maximum sample to the second minimum sample, an odd harmonic adding step is performed, wherein the odd harmonic adding step adds odd harmonics to the digital sound signal, and comprises the following steps: subtracting a fifth correction value from the first minimum sample, the fifth correction value being obtained by multiplying a fifth difference between the first minimum sample and a fifth adjacent sample by a first odd harmonic coefficient, the fifth adjacent sample being a sample preceding the first minimum sample, the first odd harmonic coefficient being selected according to a third sampling interval, the third sampling interval being an interval between the first minimum sample and a second maximum sample immediately preceding the first minimum sample; adding a sixth correction value to the first maximum sampling, wherein the sixth correction value is obtained by multiplying the second difference by a second odd harmonic coefficient, wherein the second odd harmonic coefficient is selected according to the first sampling interval; and A seventh correction value is subtracted from the second minimum sample, wherein the seventh correction value is obtained by multiplying the fourth difference by a third odd harmonic coefficient, wherein the third odd harmonic coefficient is selected according to the second sampling interval.

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