JPH11126097A - Method and device for processing voice information, and method for recording voice information on recording medium - Google Patents

Abstract

PROBLEM TO BE SOLVED: To convert a voice information of a narrow frequency band into a voice information of a wide frequency band with a small-sized, simple and inexpensive circuit constitution. SOLUTION: A harmonic generation circuit 11 compares the voice data supplied from the outside at every one sample, and detects the top peak and the under peak of the voice data based on these comparison outputs, and detects respective patterns of the comparison outputs between continuous top peaks and under peaks. Further, the circuit 11 forms the addition/subtraction data answering to harmonics according to these respective patterns, and supplies the addition/subtraction data to an adder 13 at the timing corresponding to the above respective patterns. The adder 13 addition/subtraction-processes the addition /subtraction data formed by the harmonic generation circuit 11 to the voice data supplied from the outside. Thus, a harmonic component is added to the original voice data with a small-sized, simple and inexpensive circuit constitution of only the addition/subtraction, and the voice data of the wide frequency band can be formed.

Description

DETAILED DESCRIPTION OF THE INVENTION

[0001]

BACKGROUND OF THE INVENTION 1. Field of the Invention The present invention relates to audio data recorded on, for example, a so-called compact disc (CD).
Further, it is preferably provided in a remaster device or the like that converts audio data into audio data for a video disk (digital video disk (registered trademark) or digital versatile disk: DVD (registered trademark)) that requires high sound quality and re-records (remasters). More specifically, the present invention relates to a simple audio information processing apparatus, an audio information processing method, and a method of recording audio information on a recording medium, specifically, generating harmonics based on encoded audio information, and using the generated harmonics as predetermined band components in the original audio information. The present invention relates to an audio information processing apparatus, an audio information processing method, and a method for recording audio information on a recording medium, which are intended to expand the bandwidth of original audio information by adding the information.

[0002]

2. Description of the Related Art At present, an analog input audio signal is sampled at a predetermined sampling frequency and quantized to form coded audio data with a limited band, and this is converted to an optical disk or the like. Recording on a recording medium is generally performed. As a typical recording medium on which this audio data is recorded,
A so-called compact disc (CD) is known.
This compact disk is adapted to record 16-bit audio data formed by sampling at a sampling frequency of 44.1 kHz.

[0003] Today, MPEG devices (MP
A moving image compression processing device such as EG (Moving Picture Experts Group) encodes high-efficiency audio information and image information with a large amount of information into high-efficiency compression coding, and records this on an optical disk of the same size (12 cm diameter) as a compact disk. Digital video disks (DVDs) are known and are becoming popular. In the case of this digital video disc, 24-bit (or 20-bit) audio formed by sampling an analog audio signal at a sampling frequency of 96 kHz (or 88.2 kHz expected to be newly added as a standard). The data is recorded.

For example, assuming that an analog audio signal has a frequency band up to 48 kHz as shown by a dotted line in FIG. 15, this analog audio signal is converted to a sampling frequency of 44.1 kHz for a compact disc. When the audio data is converted into 16-bit audio data, the audio data has a frequency characteristic in which a frequency band of 22.05 kHz or more has been removed as shown by a dashed line in FIG. On the other hand, when an analog audio signal is sampled at a sampling frequency of 96 kHz for a digital video disc and converted into 24-bit audio data, as shown by a solid line in FIG.
Audio data having a frequency band up to z can be formed.

Here, when an analog audio signal is converted into digital audio data, the resolution is determined by the number of quantization bits, and the frequency band is determined by the sampling frequency. For this reason, 44.1k for compact discs
16-bit audio data sampled and formed at a sampling frequency of 88.2 kHz, for example.
Even if the oversampling process is performed at the sampling frequency of z, the original audio data of 16 bits does not include the audio in the frequency band of 22.05 kHz or more. The bandwidth does not change.

In theory, the limit of human hearing is about 20 kHz
However, even if it is impossible to hear, the difference between the frequency band of the compact disc and the frequency band of the digital video disc is 2 as shown by the diagonal lines in FIG.
It is known that the presence of a sound in a frequency band of 0 kHz or more results in a richer sense of hearing.

[0007] From such a situation, a technique for shaping the waveform of the original audio information to emphasize or add harmonics to record and reproduce richer audio has been actively studied. Japanese Patent Application Laid-Open No. 127672 discloses a technique for obtaining a non-linear waveform using a conversion table.
Japanese Unexamined Patent Publication No. 75478 discloses a technique for forming a complicated nonlinear waveform by further performing a differential operation.
Japanese Patent Publication No. 6687 discloses a technique for forming a harmonic by performing non-linear processing after oversampling.
Japanese Patent Laid-Open Publication No. Hei 7-236193 discloses a technique for extracting a wideband component by performing non-linear processing after oversampling and adding this to original audio information to form wideband audio information. Have been.

[0008]

However, the prior art for emphasizing or adding harmonics uses a conversion table for non-linear processing, a differentiating circuit, or a cubic circuit, which increases the cost. In addition, there has been a problem that the circuit scale and the chip size are increased and productivity is poor. In today's world where price destruction and downsizing are required, it is an important issue how to provide a small-sized and high-performance one at low cost.

SUMMARY OF THE INVENTION The present invention has been made in view of the above problems, and has a small, simple, and inexpensive circuit configuration, and is capable of converting narrow frequency band voice information into wide frequency band voice information. It is an object of the present invention to provide an audio information recording method, an audio information recording device, and a method for recording audio information on a recording medium.

[0010]

According to the present invention, there is provided a speech information processing method comprising the steps of: comparing speech information supplied from the outside for each predetermined sample in order to solve the above-mentioned problem;
Detecting the maximum value and the minimum value of each waveform of the audio information, detecting the interval from the maximum value to the minimum value detected in the step, and detecting the interval from the minimum value to the maximum value, Forming a predetermined band component corresponding to the interval from the maximum value to the minimum value detected in the step, or the interval from the minimum value to the maximum value, based on the supplied audio information; Adding, subtracting or adding or subtracting the predetermined band component to audio information supplied from outside at a timing corresponding to the interval from the maximum value to the minimum value or the interval from the minimum value to the maximum value. By
Adding a predetermined band component to audio information supplied from the outside.

[0011] Further, the speech information processing apparatus according to the present invention comprises:
A comparison unit that compares audio information supplied from the outside for every predetermined sample in order to solve the above-described problem, a maximum and minimum detection unit that detects a maximum value and a minimum value of each waveform of the audio information, An interval between the maximum value and the minimum value detected by the detection unit, and an interval detection unit that detects an interval from the minimum value to the maximum value; and an interval detection unit that detects the interval based on audio information supplied from the outside. Band component forming means for forming a predetermined band component corresponding to the interval from the maximum value to the minimum value, or the interval from the minimum value to the maximum value, and the interval from the maximum value to the minimum value detected by the interval detection means Alternatively, at a timing corresponding to an interval from a minimum value to a maximum value, a predetermined band component formed by the band component forming means is added to audio information supplied from the outside. By subtraction or addition and subtraction, and a band component adding means for outputting adding a predetermined band component to the audio information supplied from the outside.

Such a voice information processing method and a voice information processing apparatus provide an interval between a maximum value and a minimum value of voice information,
Alternatively, a predetermined band component corresponding to the interval from the minimum value to the maximum value is formed. Then, at a timing corresponding to the interval, the predetermined band component is added to, subtracted from, or added to or subtracted from audio information supplied from the outside, thereby extending the frequency band of the audio information. As a result, it is possible to extend the frequency band by simple processing of only addition and subtraction,
It can be realized with a small, simple and inexpensive circuit configuration.

Next, a method for recording audio information on a recording medium according to the present invention is a method for recording audio information by a processing step according to any one of claims 1 to 13. Recording information on a predetermined recording medium.

According to such a method of recording audio information on a recording medium, a predetermined band component corresponding to an interval from a local maximum value to a local minimum value or an interval from a local minimum value to a local maximum value is formed. Then, at a timing corresponding to the interval, the predetermined band component is added, subtracted, or added / subtracted to audio information supplied from the outside to extend the frequency band of the audio information, and Is recorded on a predetermined recording medium. As a result, the frequency band can be extended by simple processing of only addition and subtraction, and recording can be performed on a desired recording medium, thereby realizing a small, simple, and inexpensive circuit configuration.

[0015]

DETAILED DESCRIPTION OF THE PREFERRED EMBODIMENTS Preferred embodiments of a speech information processing method, a speech information processing apparatus and a method for recording speech information on a recording medium according to the present invention will be described below in detail with reference to the drawings. An audio information processing method, an audio information processing apparatus, and a method for recording audio information on a recording medium according to the present invention are applied to a remaster device for extending the frequency band of audio data reproduced from a compact disc and re-recording the digital data on a digital video disc. Can be applied.

The remaster device according to the first embodiment of the present invention has an input terminal 1 to which 16-bit audio data sampled and formed at a sampling frequency of 44.1 kHz is supplied as shown in FIG. And this 16
A bit conversion circuit 2 for converting bit audio data into 24-bit audio data, a sampling rate conversion circuit 40 for converting a sampling frequency to 96 kHz based on the 24-bit audio data, A waveform shaping unit 3 for forming a harmonic based on the audio data and adding the same to the original 24-bit audio data to expand the frequency band; and a 24-bit audio data with the expanded frequency band. In a recording medium 6 such as a digital video disk.

As the sampling rate conversion circuit 40, any sampling frequency such as a sampling frequency of 88.2 kHz may be used as long as the sampling frequency is a sampling frequency of 44.1 kHz or higher. As an example, the sampling rate conversion circuit 40 converts the sampling frequency to 96 kHz including oversampling.

The waveform shaping unit 3 includes an I / O port 10 for taking in 24-bit audio data from the sampling rate conversion circuit 40, and a harmonic generation unit for forming harmonic data based on the 24-bit audio data. Circuit 11
A delay circuit 12 for delaying the audio data supplied via the I / O port 10 by a time corresponding to the time required for the harmonic generation processing in the harmonic generation circuit 11; It has an adder 13 for adding harmonic data from the generation circuit 11 to form audio data with an expanded frequency band, and an I / O port 14 for outputting audio data from the adder 13. .

The harmonic generation circuit 11 has the configuration shown in FIG. 2 and has an input terminal 21 to which 24-bit audio data taken in through the I / O port 10 is supplied.
A delay circuit 22 for delaying the audio data by one sample, the current audio data supplied via the input terminal 21, and the audio data delayed by one sample by the delay circuit 22 Circuit 2 for comparing the levels of
3 and the comparison output from the comparison circuit 23 to detect the top peak and the under peak of the waveform of the audio data, and to compare the top output from the top peak to the under peak and the output from the under peak to the top peak. And a peak-to-peak comparison output forming circuit 24 for outputting a comparison output.

The harmonic generation circuit 11 includes a pattern detection circuit 25 for detecting which of the plurality of data patterns stored in advance the comparison output from the peak-to-peak comparison output forming circuit 24 corresponds to. A selector 26 for selecting each pattern detection output from the pattern detection circuit 25, a shift amount control table 27 corresponding to each pattern detection output from the selector 26 and storing an addition / subtraction amount for forming a harmonic. have.

The harmonic generation circuit 11 is connected to the input terminal 2
1, a difference detection circuit 28 for detecting a difference between sample values corresponding to each pattern detection output from the selector 26, and a difference detection circuit 2 based on the audio data supplied via the selector 1.
8 for the shift amount control table 2
A bit shifter 29 that forms addition / subtraction data by performing a bit shift process corresponding to the above-mentioned addition / subtraction amount from.
And an addition / subtraction timing control circuit 30 that outputs addition / subtraction data from the bit shifter 29 via an output terminal 31 at a timing corresponding to each pattern detection output from the selector 26.

As will be described in detail below, the remaster device supplies the addition / subtraction data from the addition / subtraction timing control circuit 30 to the adder 13 shown in FIG.
By adding / subtracting the 24-bit audio data from the data, a harmonic component is added to the 24-bit audio data and recorded.

Next, the operation of the remaster device according to the first embodiment having such a configuration will be described.

FIG. 3 is a flowchart showing a series of operations in the remaster device from the formation of audio data to which harmonics are added to the recording of the audio data on a recording medium. The operation of the remaster device will be described with reference to this flowchart. First, in this flowchart, the bit rate is converted to 24 bits by the bit conversion circuit 2 and the sampling rate conversion circuit 40 converts the bit rate to 9 bits.
The process is started by supplying the audio data converted to the sampling frequency of 6 kHz to the waveform shaping unit 3, and proceeds to step S1.

The audio data supplied to the waveform shaping unit 3 is
The signal is directly supplied to the comparison circuit 23 shown in FIG.
3 is supplied. In step S1, this comparison circuit 23
Compares the current audio data with the audio data delayed by one sample by the delay circuit 22, supplies the comparison output to the peak-to-peak comparison output forming circuit 24, and proceeds to step S2.

That is, in step S1, the supplied audio data is compared with the audio data of the previous sample for each sample, and if the current audio data has a larger sample value than the audio data of the previous sample, "0" is set. Is supplied to the peak-to-peak comparison output forming circuit 24 as the comparison output when the value is small.

In this comparison, the current audio data and the audio data of the previous sample may have the same sample value. In this case, the comparison circuit 23 compares the current audio data with the audio data of the sample before and after, and if the audio data has the same sample value, compares the audio data with the audio data of the sample before and after. The comparison is performed by going back to the sample values in order. If audio data of the same sample value continues for 9 or more samples, this indicates that the data is blank. Therefore, the comparison circuit 23 continuously performs the comparison, and when there is a change, “0” is set when the audio data of the sample at the time of the change is larger than the current audio data, and “0” is set when the audio data is smaller than the current audio data. "1" is output as the comparison output.

Next, in step S2, the peak-to-peak comparison output forming circuit 24 detects each top peak and each under-peak based on the comparison output from the comparison circuit 23, and outputs the signals from the top peak to the under-peak. , And a comparison output from the under peak to the top peak is detected and supplied to the pattern detection circuit 25 to proceed to step S3.

More specifically, when the above-mentioned comparison is performed in the comparison circuit 23, a comparison output of "0" or "1" with respect to the waveform of the audio data is obtained, for example, as shown in FIG. . “0” and “1” shown in FIG.
Are arranged at one sample interval.
As can be seen from (a), when the comparative output of “0” changes to the comparative output of “1”, the audio data corresponding to the comparative output of “0” which is a sample before the comparative output of “1” is Indicates "top peak". Similarly, FIG.
As can be seen from (a), when the comparative output of "1" changes to the comparative output of "0", the audio data corresponding to the comparative output of "1" which is a sample before the comparative output of "0" is Indicates "under peak".

Therefore, the peak-to-peak comparison output forming circuit 24
Detects the top peaks A1, A2, A3,... As shown in FIG. 4A, and detects the underpeaks B1, B2, B3,. Then, the comparison output between the continuous top peak and the under peak is supplied to the pattern detection circuit 25 as a peak-to-peak comparison output. That is, the comparison output between the continuous top peak and the under peak, for example, the top peak A1 and the under peak B1, is “1,
1, the comparison output of “1, 1” is supplied to the pattern detection circuit 25 as a peak-to-peak comparison output, and the comparison output between the underpeak B1 and the top peak A2 is “0, 0, 0,. 0, 0 ”, the“ 0, 0,
The comparison output of “0, 0, 0” is supplied to the pattern detection circuit 25 as a peak-to-peak comparison output.

Next, when such a peak-to-peak comparison output is supplied to the pattern detection circuit 25, the remaster device proceeds to step S3. In step S3, each pattern stored in the pattern detection circuit 25 is compared with the peak-to-peak comparison output, and it is determined which of the previously stored patterns corresponds to this peak-to-peak comparison output.

That is, the pattern detection circuit 25 stores in advance a continuous pattern of the same peak-to-peak comparison output appearing between successive top peaks and underpeaks. Specifically, the pattern detection circuit 25 includes a “1fs pattern” indicating that the same peak-to-peak comparison output between the top peak and the underpeak is continuous for only one sample of “1” or “0”; The continuation is "1,
"2 fs pattern" indicating that it is for two samples of "1" or "0, 0", and "2 fs pattern" indicating that the continuation is for three samples of "1, 1, 1" or "0, 0, 0" 3f
s pattern "," 4 fs pattern "indicating that the continuation is for four samples of" 1, 1, 1, 1 "or" 0, 0, 0, 0 ", and the continuation is" 1, 1, 1,. 1, 1 "
Alternatively, a “5 fs pattern” indicating that it is for five samples “0, 0, 0, 0, 0”, and the continuation is “1, 1, 1,.
A “6 fs pattern” indicating that there are six samples of “1, 1, 1” or “0, 0, 0, 0, 0, 0”, and the continuation is “1, 1, 1, 1, 1, 1,. 1 ”or“ 0, 0,
A “7 fs pattern” indicating that it is for 7 samples of “0, 0, 0, 0, 0”, and the continuation is “1, 1, 1, 1, 1,.
1, 1, 1 "or" 0, 0, 0, 0, 0, 0, 0,
An “8 fs pattern” indicating eight samples of “0” and a “special pattern” indicating a continuation of “0” or “1” for nine or more samples are respectively stored in the memories 25 a to 25 a.
25i. The pattern detection circuit 25 compares each pattern stored in each of the memories 25 a to 25 i with the peak-to-peak comparison output in parallel, and supplies each comparison output to the selector 26.

The selector 26 selects, from among the comparison outputs supplied from the pattern detection circuit 25, a comparison output indicating that the pattern matches the peak-to-peak comparison output, and outputs the comparison output to the shift amount control table 27, the difference detection circuit 28 and the addition / subtraction timing control circuit 30.

Next, when the comparison output from the selector 26 is supplied to the shift amount control table 27, the difference detection circuit 28, and the addition / subtraction timing control circuit 30, the remaster device proceeds to step S4 to form a harmonic. Is detected.

That is, the remaster device adds or subtracts a level corresponding to a comparison output from the selector 26 to a predetermined sample value of 24-bit audio data (original audio data) from the delay circuit shown in FIG. Thus, the frequency band of the original audio data is extended.

More specifically, when the pattern of the comparison output between the peaks of the top peak and the under peak is not less than 2 fs pattern and not more than 5 fs pattern, the top peaks A1, A2, A3 are as shown in FIG. .. Of the sample values before and after “C1, C2, C4, C5,
.. C8, C9, C10, C11..., "Addition processing" data at a level corresponding to the difference between the top peak and the preceding and following sample values, or under peaks B1, B2, B
.. Of the sample values of 3...
3, C6, C7, C9, C10... "Subtract" data at a level corresponding to the difference between the top peak and the sample values before and after (the sample value before and after the top peak is a predetermined value). Is added, and a predetermined value is subtracted from the sample values before and after the under peak.)

When the pattern of the comparison output between the peaks of the top peak and the under peak is not less than the 6 fs pattern and not more than the 8 fs pattern, the samples of the top peaks A5, A6,. The value before and after “sample values before and after C21, C22, C29, C30 ...
. ”, The data of the level according to the difference between the top peak and the sample values before and after it is“ added ”, and the sample values of the top peaks A5, A6,. C24, C31, C32 ... "
In addition, the data of the level according to the difference between the sample value before and after the top peak and the sample value before and after the top peak is "addition processing"
(A predetermined value is added to the sample values before and after the top peak and the sample values before and after the top peak).

When the pattern of the comparison output between the peaks of the top peak and the under peak is not less than 6 fs pattern and not more than 8 fs pattern, as shown in FIG. .., The data of the level corresponding to the difference between the underpeak and the sample values before and after the underpeak is “subtracted”, and the values of the underpeaks B4, B5,. From the sample values "the sample values C27 and C28 before and after", "subtraction processing" is performed on data at a level corresponding to the difference between the sample values before and after the underpeak and the sample values immediately before and after the underpeak (underunder). A predetermined value is subtracted from the sample values before and after the peak and the sample values before and after the peak.)

As described above, each sample value is 96
Since the signal is sampled at a sampling frequency of kHz, the peak-to-peak comparison output pattern between the top peak A7 and the underpeak B6 and between the top peak A8 and the underpeak B7 shown in FIG. If the pattern of the peak-to-peak comparison output is a 1 fs pattern, it indicates that the sample value is a sufficiently high harmonic. In this case, the above-described addition / subtraction processing is not performed.

Similarly, the pattern of the comparison output between the peaks of the top peak and the under peak is the pattern of the comparison output between the peaks between the under peak B7 and the top peak A7 and between the peaks after the top peak A9 shown in FIG. When the pattern of the peak-to-peak comparison output is 9 fs pattern or more, the sample value indicates a blank or the like, and in this case, the above-described addition / subtraction processing is not performed.

The amount of addition / subtraction performed on each sample value is detected as follows.

As described above, the difference detection circuit 28 shown in FIG.
Is supplied with the comparison output from the selector 26, the difference detection circuit 28 detects a difference between sample values corresponding to each pattern indicated by the supplied comparison output.

That is, the supplied comparison output is "2fs
5A, the difference detection circuit 28 determines the difference level A between the sample value of the top peak and the sample values before and after the top peak, as shown in FIG. Alternatively, the difference level B between the sample value of the under peak and the sample values before and after the under peak is detected as shown in FIG. When the supplied comparison output indicates a pattern “6 fs pattern or more and 8 fs pattern or less”, the difference detection circuit 28
The difference level C between the sample values before and after the top peak and the sample values before and after the top peak as shown in FIG. 5C, or the sample level before and after the under peak as shown in FIG. A difference level D from the sample value immediately before and after the underpeak is detected, and this is detected by the bit shifter 2.
9.

The pattern of the peak-to-peak comparison output is 1f
In the case of the s pattern or the special pattern of 9 fs or more, the difference detection circuit 28 does not perform the detection of the difference level because the addition / subtraction process is not performed as described with reference to FIG. It has become.

On the other hand, in the shift amount control table 27, the respective difference levels A to D supplied to the bit shifter 29 are stored.
In the bit shifter 29 in order to obtain an addition / subtraction amount corresponding to each pattern of 2 fs to 8 fs.
The shift amount of D is stored in advance in, for example, four stages for each of the difference levels A to D.

More specifically, in the shift amount control table 27, as shown in FIG.
In the case of the fs pattern or the 3fs pattern, each shift amount for forming the addition / subtraction amount by setting the difference level A or the difference level B to 1/2 to 1/16 according to the level is stored. Further, the shift amount control table 27 indicates that the difference level A or the difference level B when the pattern of the peak-to-peak comparison output is the 4 fs pattern or the 5 fs pattern is set to a level of 1/4 to 1/32 according to the level. Each shift amount for forming an addition / subtraction amount is stored. Further, the shift amount control table 27 shows the difference levels A to D when the pattern of the peak-to-peak comparison output is the 6 fs pattern or the 7 fs pattern according to the level.
Each shift amount for forming an addition / subtraction amount is stored as a level of 8 to 1/64. Further, the shift amount control table 27 has a peak-to-peak comparison output pattern of 8 fs.
In the case of a pattern, each shift amount for forming an addition / subtraction amount with the difference levels A to D being 1/16 to 1/128 according to the level is stored.

The amount of addition / subtraction corresponding to each shift amount is a value found by the applicant of the present invention based on ergonomics by repeating trial listening over several years. Therefore, as will be described later, a high-quality sound output can be obtained by adding / subtracting the addition / subtraction amount to the original audio data.

The shift amount control table 27 includes a selector 2
6, a pattern of the peak-to-peak comparison output is detected based on the comparison output supplied from Step 6, and the above-described shift amount (FIG. 6) corresponding to this pattern is detected. Then, the shift data indicating the shift amount is supplied to the bit shifter 29 to which the difference levels A to D are supplied.

The bit shifter 29 performs the operation of the difference detection circuit 2 based on the shift data from the shift amount control table 27.
8 by performing a bit shift process on each of the difference levels A to D, so that each of the difference levels A to D has a level of 1/2 to 1/128 according to the pattern of the peak-to-peak comparison output. Is supplied to the addition / subtraction timing control circuit 30 as an addition / subtraction amount (addition / subtraction data).

As described above, the addition / subtraction data is added to the sample values before and after the top peak or the sample values before and after the top peak, and the sample values before and after the under peak or before and after the under peak are added. This addition / subtraction data is subtracted from the sample value of the bit shifter 2.
9 indicates “1 (addition)” or “0 (subtraction)” for the most significant bit (MSB) of the addition / subtraction data that is a sign bit.
To be specified.

Next, when the addition / subtraction data is formed as described above, the remastering apparatus proceeds to step S5, and controls the output timing of the formed addition / subtraction data.

That is, as described above, when the addition / subtraction data is added / subtracted to the original audio data, the timing of the sample value before and after the top peak or under peak when the pattern of the peak-to-peak comparison output is a 2 fs pattern to 5 fs pattern. At the timing of adding / subtracting the addition / subtraction data.
In the case of the s pattern to the 8 fs pattern, it is the timing to add / subtract the addition / subtraction data to the sample values before and after the top peak or the under peak and the sample values before and after the top peak or the under peak.

Therefore, in step S5,
The addition / subtraction timing control circuit 30 measures the output timing of the addition / subtraction data from the bit shifter 29 in accordance with the pattern of the peak-to-peak comparison output indicated by the comparison output from the selector 26, and outputs the addition data via the output terminal 31 at each of the above timings. To the adder 13 shown in FIG. Thus, when the pattern of the peak-to-peak comparison output is the 2 fs pattern to the 5 fs pattern, the sample values before and after the top peak or under peak are supplied to the adder 13 as shown by the dotted arrows in FIG. At the timing, the addition / subtraction data is supplied to the adder 13. When the pattern of the peak-to-peak comparison output is the 6 fs pattern to the 8 fs pattern, the sample values before and after the top peak or the under peak and before and after the top peak or the under peak as shown by the dotted arrows in FIG. Addition / subtraction data is supplied to the adder 13 at a timing when the sample value is supplied to the adder 13 later.

Next, when the addition / subtraction data is supplied to the adder 13, the remaster device proceeds to step S6. In step S6, the adder 13 performs an addition / subtraction process on the original 24-bit audio data supplied from the delay circuit 12 with the addition / subtraction data (harmonics) formed by the harmonic generation circuit 11. FIGS. 7A to 7C and FIGS. 8A to 8C show specific examples of waveform changes before and after the addition / subtraction processing.

FIGS. 7A to 7C show the addition processing when the pattern of the peak-to-peak comparison output is the 2 fs pattern. As can be seen from FIGS. 7A to 7C, when the pattern of the peak-to-peak comparison output is the 2 fs pattern, the difference between the sample value of the top peak of the original audio data and the sample values before and after the top peak is obtained. The addition / subtraction data calculated according to the level is added to the sample values before and after the top peak, as indicated by oblique lines in FIGS. Thereby, the waveform of the original audio data shown by the middle dotted line in FIGS. 7A to 7C is shaped into a form in which the frequency band is extended as shown by the solid line in FIGS. 7A to 7C. be able to.

Similarly, FIGS. 8A to 8C show the addition / subtraction processing when the pattern of the peak-to-peak comparison output is the 3 fs pattern. As can be seen from FIGS. 8A to 8C, when the pattern of the peak-to-peak comparison output is a 3 fs pattern, the difference between the sample value of the top peak of the original audio data and the sample values before and after the top peak is obtained. The addition / subtraction data calculated according to the level is shown in FIG.
As shown by the right diagonal line in (c), addition processing is performed on the sample values before and after the top peak. Also, the addition and subtraction data calculated according to the difference level between the sample values before and after the underpeak of the original audio data and the sample values before and after the underpeak are shown in FIGS. As indicated by oblique lines, addition processing is performed on sample values before and after the top peak. Thereby, the waveform of the original audio data shown by the middle dotted line in FIGS. 7A to 7C is shaped into a form in which the frequency band is extended as shown by the solid line in FIGS. 7A to 7C. be able to.

The 24-bit audio data that has undergone the waveform shaping process for expanding the frequency band (adding higher harmonics) is supplied to the recording system 5 via the I / O port 14.
In the remastering device, the 24-bit audio data subjected to the waveform shaping process is supplied to the recording system 5 via the I / O port 14, and the process proceeds to step S7. The output audio data may be subjected to a rounding process and output as, for example, 20-bit audio data.

In step S7, the recording system 5 records the 24-bit audio data subjected to the waveform shaping process on a recording medium 6 such as a digital video disk. Then, after confirming the recording of all the audio data, the entire routine of the flowchart shown in FIG. 3 ends. This makes it possible to re-record (remaster) the digital video disk after adding the insufficient harmonic components indicated by oblique lines in FIG. 15 to the 16-bit audio data reproduced from the compact disk.

As is apparent from the above description, the remastering apparatus according to the first embodiment of the present invention converts audio data for a compact disk into audio data for a digital video disk, and converts the audio data for the digital video disk to the digital video disk. Can be re-recorded. Therefore, it is possible to omit the trouble of newly forming the audio data for the digital video disc from the analog audio signal, and it is possible to reuse the existing audio data for the compact disc.

The waveform shaping section 3 can perform waveform shaping of the original audio data (formation of a harmonic component and synthesis of the harmonic component with the original audio data) using only addition and subtraction processing. it can. For this reason, a conversion table for nonlinear processing conventionally required for such waveform shaping,
The waveform shaping can be performed without using a differentiating circuit or a cubic circuit, and a high-pass between the harmonic generating circuit 11 and the adder 13 for extracting a harmonic component from a nonlinear signal. The need for providing a filter (HPF) can be eliminated. In particular, a filter system such as a high-pass filter, as can be seen from the configuration of a so-called IIR filter or FIR filter, causes an increase in circuit size and complexity when trying to obtain good filter characteristics. The use of a filter causes waveform distortion, but the waveform shaping unit 3 can fundamentally remove this factor. Therefore, the circuit size of the waveform shaping unit 3 can be reduced to reduce the chip size, and cost reduction, productivity improvement, and high performance can be achieved. In addition, since a small-sized and high-performance device can be provided at a low price, it is possible to sufficiently cope with today's price destruction and downsizing.

Next, a remaster device according to a second embodiment of the present invention will be described. The remaster device according to the first embodiment has a sampling frequency of 44.1 kHz.
The data processing is performed by converting the sampling rate and the bit rate of the audio data of bits into 24-bit audio data at a sampling frequency of 96 kHz, for example. For example, 16-bit audio data of a sampling frequency of 96 kHz is supplied, and only the bit rate is converted from 16 bits to 24 bits to perform data processing while the sampling frequency is not changed.

The remaster device of the second embodiment differs from the remaster device of the first embodiment only in this point. For this reason, in the following description, only this difference will be described, and portions showing the same operation as the remaster device of the first embodiment will be denoted by the same reference numerals in FIG. The explanation will be avoided.

That is, in the remaster device according to the second embodiment, the sampling rate conversion circuit 40 connected between the bit conversion circuit 2 and the waveform shaping section 3 as shown in FIG. 9 is omitted. It has the form of a shape.

In such a remaster device, audio data having a sampling frequency of 96 kHz and a bit rate of 16 bits supplied via the input terminal 1 is converted into 24-bit audio data by the bit conversion circuit 2 and is subjected to waveform shaping. Supply to section 3.

The pattern detection circuit 25 provided in the waveform shaping section 3 of the remaster device according to the first embodiment described above.
In the remaster apparatus of the second embodiment, the pattern detection circuit 25 provided in the waveform shaping unit 3 stores the 1 fs pattern to 8 fs pattern and the special pattern. pattern~
In addition to the 8 fs pattern, the continuation of the same peak-to-peak comparison output between the top peak and the under peak is “1, 1,
1, 1, 1, 1, 1, 1, 1 "or" 0, 0, 0,
A “9fs pattern” indicating that it is for 9 samples of “0, 0, 0, 0, 0, 0”, and the continuation is “1, 1, 1, 1,.
1, 1, 1, 1, 1, 1 "or" 0, 0, 0, 0,
“10 fs pattern” indicating 10 samples of “0, 0, 0, 0, 0, 0” and “special pattern” indicating a continuation of “0” or “1” for 11 samples or more
Are respectively stored.

The difference detection circuit 28 shown in FIG.
fs pattern to 3 fs pattern;
In the case of a “special pattern” indicating a continuation of “0” or “1” for one sample or more, the above-described detection of the difference level is not performed.

The shift amount control table 27 includes a selector 2
When the pattern of the peak-to-peak comparison output detected in step 6 is the 4 fs pattern to the 10 fs pattern, the driving is performed. When the pattern of the peak-to-peak comparison output is the 4 fs pattern or the 5 fs pattern, as shown in FIG. The difference level A or the difference level B described above (see FIG. 5)
Controls the bit shifter 29 so that the level becomes 1/2 to 1/16 according to the level. Selector 2
When the pattern of the peak-to-peak comparison output detected in step 6 is a 6 fs pattern or a 7 fs pattern, the difference level A or the difference level B becomes 1 /
The bit shifter 29 is controlled to have a level of 4 to 1/32. When the pattern of the peak-to-peak comparison output detected by the selector 26 is an 8 fs pattern or a 9 fs pattern, the above-described difference levels A to D (see FIG. 5).
Controls the bit shifter 29 so that the level becomes 1/8 to 1/64 according to the level. Further, the pattern of the peak-to-peak comparison output detected by the selector 26 is 10 f
In the case of the s pattern, the bit shifter 29 is controlled so that the difference levels A to D become 1/16 to 1/128 according to the levels.

Thus, it is possible to emphasize the higher harmonics of the treble in the frequency band of the audio data to be output or recorded, and it is possible to obtain the same effects as those of the remaster device of the first embodiment.

Next, a description will be given of a remaster device according to a third embodiment of the present invention. The remaster device according to the third embodiment has a low-pass filter 41 provided between the waveform shaping section 3 and the recording system 5 of the remaster device according to the second embodiment as shown in FIG. is there. It should be noted that the remastering device of the third embodiment has only this point in the first embodiment.
Since this is different from the remaster device of the embodiment, only the difference will be described in the following description.

The effect of the remaster device of the first embodiment is that the low-pass filter can be omitted. One of the effects is that if aliasing noise or the like occurs, Aliasing noise and the like can be removed by the low-pass filter 41, and high sound quality of the audio data to be formed can be secured.

It is to be understood that the remaster device does not necessarily require a low-pass filter, and that such a remaster device can cope with an aliasing noise or the like in the unlikely event that it is provided.

Next, a fourth embodiment of the present invention will be described. In the fourth embodiment, the audio information processing method and the audio information processing apparatus according to the present invention are applied to an audio processing apparatus for a CD player. In the description of the audio processing apparatus for a CD player according to the fourth embodiment, the same reference numerals in FIG. 12 denote the same parts as those in the first embodiment, and the detailed description thereof will be omitted. Description is omitted.

That is, in the audio processing apparatus for a CD player according to the fourth embodiment, audio data output as digital data to the subsequent stage of the waveform shaping section 3 is converted into an analog audio signal as shown in FIG. It has a configuration having a D / A converter 50, and this D / A converter 50
The 24-bit audio data to which the harmonics are added is converted into an analog signal, which is supplied to a recording device via an output terminal 60, for example, to a speaker device or an optical disk recording device. As described above, since the audio data formed by the waveform shaping unit 3 has an extended frequency band, when the audio data is converted into an analog signal and supplied to the speaker device,
In addition to obtaining rich sound effects, the same effects as those of the above embodiments can be obtained.

Next, a fifth embodiment of the present invention will be described. In the fifth embodiment, the audio information processing method and the audio information processing apparatus according to the present invention are applied to an audio processing apparatus for a DVD player. In the description of the audio processing apparatus for a CD player according to the fifth embodiment, the same reference numerals in FIG. 13 denote parts showing the same operations as those in the first embodiment, and the detailed description thereof will be omitted. Description is omitted.

That is, the DVD of the fifth embodiment
As shown in FIG. 13, an audio processing device for a player includes an oversampling circuit 40 for sampling audio data having a sampling frequency of 96 kHz and a bit rate of 24 bits, a sampling frequency of 192 kHz without changing the bit rate, and this oversampling circuit. The sampling frequency supplied from the input terminal 1 is audio data having a sampling frequency of 192 kHz from 40 and a sampling frequency of 96.
a switch 65 for switching and outputting audio data of kHz, and a D / A for converting audio data having a sampling frequency of 96 kHz output from the waveform shaping unit 3 and audio data having a sampling frequency of 192 kHz into analog.
And a conversion circuit 50.

The 1fs pattern to 8fs pattern and the special pattern described with reference to FIG.
1 fs pattern to 10 fs described with reference to FIG. 10 for audio data having a sampling frequency of kHz.
A pattern and a special pattern are respectively stored for the audio data of the sampling frequency of 96 kHz.

The controller 70 controls the switching of the changeover switch 65 in accordance with the sampling frequency of the audio data to be processed by the audio processing device, and controls the pattern used in the pattern detection circuit 25 of the high frequency generation circuit 11. , D / A conversion circuit 50
The driving frequency is controlled by switching.

Next, the operation of the audio processing apparatus for a DVD player according to the fifth embodiment having such a configuration will be described.

First, when performing data processing of audio data having a sampling frequency of 96 kHz, the controller 70
10 controls the changeover switch 65 so that the selected terminal 65a is selected by the selection terminal 65c.
While controlling the high-frequency generation circuit 11 so as to perform the above-described peak-to-peak comparison output pattern detection with the 1fs pattern to 10fs pattern and the special pattern described using
The D / A conversion circuit 50 is controlled to perform a D / A conversion process corresponding to a sampling frequency of 96 kHz.

Thus, 96 kHz from the input terminal 1
Is supplied to the waveform shaping unit 3 via the changeover switch 65, and a harmonic component corresponding to a sampling frequency of 96 kHz is added thereto, and is converted into an analog signal by the D / A conversion circuit 50. For example, the speaker device or the optical disk recording device And so on.

The sampling frequency is 192 kHz.
When the data processing of the audio data of
0 controls the changeover switch 65 so that the selected terminal 65b is selected by the selection terminal 65c.
The high frequency generation circuit 11 is controlled so as to perform the above-described peak-to-peak comparison output pattern detection using the 1fs pattern to 8fs pattern and the special pattern described with reference to FIG.
The D / A conversion circuit 50 is controlled so as to perform a D / A conversion process corresponding to a sampling frequency of 92 kHz.

Thus, 96 kHz from the input terminal 1
Is oversampled by the oversampling circuit 40.
The signal is converted to a sampling frequency of 92 kHz and supplied to the waveform shaping unit 3 via the changeover switch 65. Then, a harmonic component corresponding to the sampling frequency of 192 kHz is added, converted into an analog signal by the D / A conversion circuit 50, and output to, for example, a speaker device or an optical disk recording device.

As described above, the DV of the fifth embodiment is described.
The audio processing device for the D player may output the sound data of 96 kHz with a harmonic component added thereto, or convert the sound data of 96 kHz to a sampling frequency of 192 kHz and then add the harmonic component and output. it can. For this reason, as shown in FIG. 14, when data processing for adding a harmonic component to audio data of 96 kHz is performed on audio data of a DVD indicated by a dashed line in the drawing,
As shown by the dotted line in the figure, it is possible to emphasize the high band, and when the data processing of adding the harmonic component after converting the 96 kHz audio data to the 192 kHz sampling frequency is performed. As shown by the solid line in the figure, further enhancement of a higher band up to 192 kHz can be achieved. Therefore, when the audio signal subjected to the data processing is supplied to the speaker device, music and the like can be enjoyed with a richer sense, and the same effects as those of the above-described embodiments can be obtained.

In the description of the fifth embodiment,
Oversampling circuit 40 and changeover switch 6
5, the data processing of the audio data of the sampling frequency of 96 kHz and the data processing of the audio data of the sampling frequency of 192 kHz can be selected. This is a configuration in which the oversampling circuit 40 and the changeover switch 65 are omitted. Is also good. As a result, the patterns stored in the pattern detection circuit 25 of the high-frequency generation circuit 11 can be changed to the 1fs pattern to 10f described with reference to FIG.
s pattern and special pattern only, and the driving frequency in the D / A conversion circuit 50 is set to 96 kHz.
Only the drive frequency corresponding to z can be set, and the above-mentioned high band can be emphasized after simplifying the configuration.

Finally, in the description of the above embodiments,
An audio information processing apparatus, an audio information processing method, and a method for recording audio information on a recording medium according to the present invention are provided by converting a compact disk audio data into a digital video disk audio data and re-recording the audio data. The present invention is applied to an audio processing device for a player or an audio processing device for a DVD player. However, the present invention is also applicable to a case in which audio data for a compact disc is converted to digital audio tape (DAT) having a sampling frequency of 48 kHz. The present invention can be applied to any device that converts audio data in a narrow frequency band into audio data in a wide frequency band, such as application to a remaster device that converts audio data to re-record.

In the description of each of the above embodiments, the sampling frequency is 44.1 kHz, 96 kHz, 19 kHz.
Although the description has been given with specific numerical values such as 2 kHz and a bit rate of audio data of 16 bits, 24 bits, etc., this is only an example for explaining the embodiment of the present invention more easily. . For this reason, the present invention is not limited to such specific numerical values or exemplary embodiments, and various modifications may be made according to design and the like within a range not departing from the technical idea of the present invention. Is of course possible.

[0087]

The voice information processing method and the voice information processing apparatus according to the present invention can convert narrow frequency band voice information into wide frequency band voice information with a small, simple and inexpensive circuit configuration. can do.

The method of recording audio information on a recording medium according to the present invention is a small, simple and inexpensive circuit configuration, which converts audio information in a narrow frequency band into audio information in a wide frequency band and stores the converted audio information in a recording medium. Can be recorded.

[Brief description of the drawings]

FIG. 1 is a first example to which a voice information processing method, a voice information processing apparatus, and a voice information recording method according to the present invention are applied.
It is a block diagram of a remastering device of an embodiment.

FIG. 2 is a block diagram of a harmonic generation circuit provided in the remaster device according to the first embodiment.

FIG. 3 is a flowchart for explaining a recording operation of the remaster device according to the first embodiment.

FIG. 4 is a diagram for explaining the timing of adding / subtracting a predetermined addition / subtraction amount for each data pattern of audio data when expanding a frequency band in the remaster device according to the first embodiment.

FIG. 5 is a diagram for explaining how to calculate the addition / subtraction amount.

FIG. 6 is a diagram illustrating an example of an addition / subtraction amount calculated for each data pattern of audio data.

FIG. 7 is a diagram showing an example of a waveform of audio data in which the addition / subtraction amount has been subjected to an addition / subtraction process.

FIG. 8 is a diagram illustrating an example of a waveform of audio data in which the addition / subtraction amount has been subjected to addition / subtraction processing.

FIG. 9 is a diagram illustrating a second example to which a voice information processing method, a voice information processing apparatus, and a method of recording voice information on a recording medium according to the present invention are applied.
It is a block diagram of a remastering device of an embodiment.

FIG. 10 is a diagram illustrating an example of an addition / subtraction amount calculated for each data pattern of audio data in the remaster device according to the second embodiment.

FIG. 11 is a block diagram of a remastering apparatus according to a third embodiment to which a voice information processing method, a voice information processing apparatus, and a method of recording voice information on a recording medium according to the present invention are applied.

FIG. 12 is a block diagram of a sound processing device for a CD player according to a fourth embodiment to which a sound information processing method and a sound information processing device according to the present invention are applied;

FIG. 13 is a block diagram of a sound processing device for a DVD player according to a fifth embodiment to which a sound information processing method and a sound information processing device according to the present invention are applied;

FIG. 14 is a diagram showing a frequency band of audio data to which harmonics are added by the audio processing device for a DVD player according to the fourth embodiment.

FIG. 15 is a diagram for explaining each frequency band of an analog audio signal, audio data of a compact disc, and audio data of a digital video disc.

[Explanation of symbols]

2 ... bit conversion circuit, 3 ... waveform shaping section, 5 ... recording system, 6
... Recording medium 10, 14 ... I / O port, 11 ... Harmonic generation circuit, 1
2 delay circuit 13 adder 22 delay circuit 23 comparison circuit 26
... Selector 24 ... Peak comparison output forming circuit, 25 ... Pattern detecting circuit 27 ... Shift amount control table, 28 ... Difference detecting circuit, 2
9 Bit shifter 30 Addition / subtraction timing control circuit 40 Sampling rate conversion circuit 41 Low pass filter 50 D / A conversion circuit 65
… Changeover switch 70… Controller

Claims (27)

[Claims] 1. a step of comparing audio information supplied from the outside for every predetermined sample; a step of detecting a maximum value and a minimum value of each waveform of the audio information; and a minimum value from the maximum value detected in the step. Detecting the interval between the maximum value and the interval from the minimum value to the maximum value, based on the audio information supplied from the outside, the interval from the maximum value to the minimum value detected in the step, or from the minimum value Forming a predetermined band component corresponding to the interval from the local maximum value to the external source at a timing corresponding to the interval from the local maximum value to the local minimum value, or the interval from the local minimum value to the local maximum value detected in the step. By adding, subtracting, or adding / subtracting the predetermined band component to the audio information to be output, a predetermined Speech information processing method and a step of adding the frequency component. 2. The method according to claim 1, wherein the step of comparing the audio information supplied from the outside for each predetermined sample includes comparing the audio information supplied from the outside for each sample at a sampling frequency of the audio information. Claim 1
The voice information processing method described in the above. 3. The step of detecting a local maximum value and a local minimum value of each waveform of the audio information is performed based on each comparison output obtained by comparing the audio information supplied from the outside for each predetermined sample. 3. The voice information processing method according to claim 1, wherein the local maximum value and the local minimum value are detected. 4. The step of detecting the maximum value and the minimum value of each waveform of the audio information comprises the step of comparing the audio information supplied from the outside for each predetermined sample. 4. The method according to claim 1, wherein a continuation of the comparison output is detected, and a sample value corresponding to the comparison output immediately before the transition of the same comparison output is detected as a maximum value or a minimum value. The speech information processing method according to any one of the preceding claims. 5. The step of detecting the interval from the local maximum value to the local minimum value and the interval from the local minimum value to the local maximum value is performed between the local maximum value and the local minimum value or between the local minimum value and the local maximum value. Detecting the intervals at which the same comparison output obtained in the step of comparing the audio information supplied from the outside for each predetermined sample is continuous, thereby detecting the interval from the local maximum value to the local minimum value, and the local minimum value to the local maximum value. The voice information processing method according to any one of claims 1 to 4, wherein an interval up to is detected. 6. A predetermined band component corresponding to an interval from a maximum value to a minimum value or an interval from a minimum value to a maximum value detected in the step is formed based on the audio information supplied from the outside. 6. The method according to claim 1, wherein the step forms a high-frequency component corresponding to an interval from the maximum value to the minimum value or an interval from the minimum value to the maximum value. The voice information processing method described in the above. 7. The step of forming a predetermined band component corresponding to the interval from the maximum value to the minimum value or the interval from the minimum value to the maximum value based on the audio information supplied from the outside, The difference between the maximum value and the previous or subsequent sample value, or the minimum value and the previous or subsequent sample value, corresponding to the interval from the value to the minimum value, or the interval from the minimum value to the maximum value Difference value between the previous sample value of the maximum value and the sample value before the maximum value or the sample value after the maximum value, or the sample value before the minimum value and the sample value before the minimum value or A predetermined band component is formed by detecting a difference value from the sample value after the minimum value and setting the difference value to a value corresponding to the level of the corresponding maximum value or minimum value. The voice information processing method according to claim 1. 8. A process for adding and subtracting the predetermined band component to audio information supplied from outside at a timing corresponding to an interval from the maximum value to the minimum value or an interval from the minimum value to the maximum value. Alternatively, the step of adding a predetermined band component to the audio information supplied from the outside by performing addition and subtraction processing includes, for a waveform corresponding to the maximum value, an interval from the maximum value to the minimum value, or a minimum value. Depending on the interval from the maximum value, the predetermined band component is added to the sample values before and after the maximum value, or the sample values before and after the maximum value, and the waveform corresponding to the minimum value is calculated. The interval from the maximum value to the minimum value,
Alternatively, the predetermined band component is subtracted from a sample value before and after the minimum value or a sample value before and after the minimum value according to an interval from the minimum value to the maximum value. The voice information processing method according to any one of claims 1 to 7. 9. A process for adding and subtracting the predetermined band component to audio information supplied from the outside at a timing corresponding to the interval from the maximum value to the minimum value or the interval from the minimum value to the maximum value. Alternatively, the step of adding a predetermined band component to the audio information supplied from the outside by performing addition and subtraction processing is such that the interval from the maximum value to the minimum value or the interval from the minimum value to the maximum value is equal to or less than the predetermined interval, 9. The method according to claim 1, wherein when the predetermined time interval is equal to or longer than the predetermined time interval, the audio information is output as it is without performing the addition processing, the subtraction processing or the addition / subtraction processing of the predetermined band component. The speech information processing method according to claim 1. 10. A process of adding and subtracting the predetermined band component to audio information supplied from the outside at a timing corresponding to an interval from the maximum value to the minimum value or an interval from the minimum value to the maximum value. Alternatively, the step of adding a predetermined band component to the audio information supplied from the outside by performing addition and subtraction processing is performed in such a manner that the interval from the maximum value to the minimum value or the interval from the minimum value to the maximum value is one sample interval or less. Alternatively, if the interval is equal to or longer than 9 sample intervals, the audio information is output as it is without performing the addition, subtraction, or addition / subtraction processing of the predetermined band component. The voice information processing method according to claim 1. 11. A step of increasing a sampling frequency for increasing the number of samples of the externally supplied audio information as a step preceding the step of comparing the externally supplied audio information for each predetermined sample. The voice information processing method according to any one of claims 1 to 10, further comprising: 12. As a step preceding the step of comparing the audio information supplied from the outside for every predetermined sample, a step of performing oversampling to make the audio information supplied from the outside twice the number of samples is provided. The voice information processing method according to any one of claims 1 to 11, wherein: 13. The method according to claim 1, further comprising a step of removing a predetermined unnecessary band component as a step subsequent to the step of adding a predetermined band component to the audio information supplied from the outside. Item 13. The audio information processing method according to any one of Items 12. 14. A comparison means for comparing audio information supplied from the outside for each predetermined sample, a maximum and minimum detection means for detecting a maximum value and a minimum value of each waveform of the audio information, and a maximum and minimum detection means. Interval detecting means for detecting an interval from the detected maximum value to the minimum value, and an interval from the minimum value to the maximum value; and a local maximum value detected by the interval detecting means based on audio information supplied from the outside. A band component forming unit that forms a predetermined band component corresponding to an interval from the minimum value to the minimum value or an interval from the minimum value to the maximum value; and an interval from the maximum value to the minimum value detected by the interval detection unit, or a minimum value. At a timing corresponding to the interval from the value to the maximum value, the predetermined band component formed by the band component forming means is added to the audio information supplied from the outside, and subtracted. By physical or addition and subtraction processing,
And a band component adding means for adding a predetermined band component to audio information supplied from the outside and outputting the added band component. 15. The audio information processing apparatus according to claim 14, wherein said comparing means compares audio information supplied from the outside for each sample at a sampling frequency of the audio information. 16. The method according to claim 14, wherein the maximum and minimum detection means detects the maximum value and the minimum value based on each comparison output obtained from the comparison means. Voice information processing device. 17. The maximum / minimum detection means detects a continuation of the same comparison output from each of the comparison outputs from the comparison means, and corresponds to a comparison output immediately before a transition of the same continuous comparison output. 17. The audio information processing apparatus according to claim 14, wherein a sample value is detected as a local maximum value or a local minimum value. 18. The interval detecting means detects an interval from the maximum value to the minimum value and an interval from the minimum value to the maximum value by detecting an interval at which the same comparison output from the comparing means continues. 2. The method according to claim 1, wherein
The speech information processing apparatus according to any one of claims 4 to 17. 19. The apparatus according to claim 19, wherein the band component forming unit includes an interval from a maximum value to a minimum value detected by the interval detection unit;
21. A high-frequency component corresponding to an interval from a minimum value to a maximum value is formed.
The voice information processing apparatus according to any one of claims 1 to 4. 20. The band component forming means includes a maximum value and a preceding sample value corresponding to an interval from a maximum value to a minimum value or an interval from a minimum value to a maximum value detected by the interval detection means. Alternatively, the difference value between the post-sample value or the difference value between the minimum value and the previous sample value or the post-sample value, and further, the pre-sample value of the pre-maximum value and the pre-sample value or the post-maximum sample value of the maximum value Or a difference value between a sample value before the minimum value and a sample value before the minimum value or a sample value after the minimum value, and a difference detected by the difference value detection means. 20. A level control means for forming a predetermined band component as a value corresponding to a level of a corresponding maximum value or minimum value, the level control means comprising: Voice information processing apparatus according. 21. The band component adding means, for the waveform corresponding to the maximum value, sets the interval from the maximum value to the minimum value detected by the interval detection means or the interval from the minimum value to the maximum value. Accordingly, a predetermined band component from the band component forming means is added to a sample value before and after the local maximum value or a sample value before and after the local maximum value, and a waveform corresponding to the local minimum value is calculated. The predetermined value is determined based on an interval from the local maximum value to the local minimum value, or a sample value before and after the local minimum value, or a sample value immediately before and after the local minimum value according to an interval from the local minimum value to the local maximum value. 21. The audio information processing apparatus according to claim 14, wherein a band component is subtracted. 22. The band component adding means, wherein the interval from the maximum value to the minimum value detected by the interval detection means,
Alternatively, the interval from the minimum value to the maximum value is equal to or less than a predetermined interval,
Alternatively, if it is equal to or longer than a predetermined interval, the audio information is output as it is without performing the addition processing, subtraction processing or addition / subtraction processing of the predetermined band component.
22. The audio information processing device according to claim 21. 23. The band component adding means, comprising: an interval from a maximum value to a minimum value detected by the interval detection means;
Alternatively, when the interval from the minimum value to the maximum value is equal to or less than one sample interval or equal to or more than nine sample intervals, the audio information is output as it is without performing the addition, subtraction, or addition / subtraction of the predetermined band component. The speech information processing apparatus according to any one of claims 14 to 22, wherein: 24. Sampling frequency increasing means for increasing the sampling frequency for increasing the number of samples of audio information supplied from the outside, and supplying this audio information to said comparing means, difference detecting means and band component adding means. The voice information processing apparatus according to any one of claims 14 to 23, comprising: 25. An oversampling means for performing oversampling for making the audio information supplied from the outside twice as many as the number of samples, and supplying this audio information to said comparing means, difference detecting means and band component adding means. The voice information processing apparatus according to any one of claims 14 to 24, wherein: 26. An apparatus according to claim 14, further comprising an unnecessary band component removing unit for removing a predetermined unnecessary band component from the audio information from said band component adding unit.
The voice information processing apparatus according to any one of claims 1 to 4. 27. The method according to claim 1, wherein
A method for recording audio information on a recording medium, comprising recording the audio information on a predetermined recording medium, comprising the processing step according to any one of the preceding claims.