JPH09120646A - Audio signal-compressing/recording apparatus and audio signal-compressing apparatus and optical recording medium - Google Patents

Abstract

PROBLEM TO BE SOLVED: To record data to CD-ROMs or DVDs by formatting compressed quantized data to a user area by a formatting means, recording the data to a recording medium and compressing audio signals with a high compression rate. SOLUTION: An A/D converter 1 quantizes audio signals input to an input terminal IN with a frequency of not lower than 441kHz. The data are written to a memory 3 via a signal-processing circuit 2. The signal-processing circuit 3 sequentially writes fresh frames normalized quantized per band to the memory 3. A CD-ROM-coding circuit 4 reads out the data and adds a synchronous signal, a header, a sub header and the like to each sector. The data are input to a CD-coding circuit 5, formatted in a CD format and recorded by a recording head via a second output terminal OUT2.

Description

Detailed Description of the Invention

[0001]

BACKGROUND OF THE INVENTION 1. Field of the Invention The present invention relates to an audio signal compression recording apparatus for high efficiency encoding an audio signal and recording it on an optical recording medium, and an audio signal compression apparatus therefor.
Furthermore, it relates to an optical recording medium.

[0002]

2. Description of the Related Art CD (Compact Disc) is 1982
Ten years have passed since it first appeared in the year, and now it has become established as a digital storage medium due to various developments. Considering the use of audio media, this media with sampling frequency fs = 44.1 kHz and quantization bit rate = 16 bits is completely in its maturity stage. further,
High density discs called DVDs are about to be used as digital discs for data in computers and the like. A digital disc is a CD or CD-RO.
An optical disc such as an M or a DVD in which audio or video signals are recorded as digital signals is referred to.

[0003]

An ordinary music (audio) CD (hereinafter referred to as CD-DA) records a two-channel audio signal with a sampling frequency fs = 44.1 kHz and a quantization bit number = 16 bits. However, the same data amount cannot be recorded in the CD-ROM format according to the conventional CD-DA standard. This is because the format of the CD-ROM has a header including a synchronization signal (SYNC), an address, and a mode, so that the recording capacity for recording the audio signal is smaller than that of the CD-DA. On the other hand, with the development and rapid spread of personal computers and their peripheral devices, there is a demand to enjoy music with high sound quality through a CD-ROM drive.

Further, in a digital disc called a DVD, sound is recorded without being compressed by the linear PCM, so that a data amount is required for recording with high hi-finess and the recording time is shortened. For this disc, data is processed by orthogonal transform and / or Huffman code to perform compression for reducing the data amount, and DV is used.
A recording device for recording in the D format and an optical disc recorded by such a method are conceivable.

Therefore, the present invention is obtained by quantizing at a quantization bit number of 16 bits and a sampling frequency of 44.1 kHz or higher due to a higher compression rate than the current data processing for CD-DA. An object is to provide an audio signal compression recording device, an audio signal compression device, and an optical recording medium capable of recording data on a CD-ROM.

Further, the present invention provides an audio signal compression recording apparatus and an audio signal compression apparatus capable of recording on a DVD audio which is generally considered to have a different data format within the range of CD variation (size / modulation method). It is an object to provide an optical recording medium.

[0007]

In order to achieve the above object, the present invention quantizes an audio signal at a quantization bit number of 16 bits and a sampling frequency of 44.1 kHz or higher and quantizes it. Orthogonal transformation is applied to each fixed amount of quantized data to reduce / compress the amount of data, and the compressed data is converted into the CD-ROMXA standard mode 2, form 2
The user data area, the user data area of the mode 2 of the CD-ROM standard, or the user data area of the DVD is formatted, and the formatted data is recorded on the recording medium.

That is, according to the present invention, the audio signal is quantized with 16 bits and the sampling frequency is 44.1.
a quantizing means for quantizing at a frequency of kHz or higher; a data compressing means for compressing the data quantity by applying orthogonal transformation for each predetermined quantity of quantized data quantized by the quantizing means; Audio signal compression recording having formatting means for formatting the data compressed by the compression means so as to be arranged in the user data area of the digital disc, and means for recording the data formatted by the formatting means in a recording medium as a CD format. A device is provided.

According to the present invention, the audio signal is quantized with 16 bits and the sampling frequency is 44.1 kHz.
Or a quantizing means for quantizing at a frequency higher than that, a data compressing means for compressing the data quantity by applying orthogonal transformation for each predetermined quantity of quantized data quantized by the quantizing means, and the data compressing means. CD-R data compressed by means
An audio signal compression recording apparatus having formatting means for formatting so as to be arranged in a user data area of mode 2 of the OMXA standard, and means for recording the data formatted by the formatting means as a CD format on a recording medium. Provided.

Further, according to the present invention, the audio signal is quantized with 16 bits and the sampling frequency is 44.1k.
A quantizing means for quantizing at a frequency of Hz or higher;
A data compression unit that applies an orthogonal transformation to each predetermined amount of quantized data quantized by the quantization unit to compress the data amount; and a data compressed by the data compression unit on a CD-
There is provided an audio signal compression recording apparatus having formatting means for formatting so as to be arranged in a user data area of ROM standard mode 2, and means for recording the data formatted by the formatting means on a recording medium as a CD format. .

Further, according to the present invention, the audio signal is quantized with 16 bits and the sampling frequency is 44.1 kHz.
a quantizing means for quantizing at a frequency of z or higher; a data compressing means for compressing the data quantity by applying orthogonal transformation for each predetermined quantity of quantized data quantized by the quantizing means; Audio signal compression recording apparatus having formatting means for formatting the data compressed by the compression means so as to be arranged in the user data area of the DVD, and means for recording the data formatted by the formatting means as a CD format on a recording medium. Will be provided.

Although the present invention can be regarded as an audio signal compression recording apparatus as described above, in order to manufacture a read-only disc, the step of recording as a CD format is a task of the disc manufacturing factory side. . Therefore, the present invention can be regarded as an audio signal compression device.

That is, according to the present invention, the audio signal is quantized with 16 bits and the sampling frequency is 44.1.
a quantizing means for quantizing at a frequency of kHz or higher; a data compressing means for compressing the data quantity by applying orthogonal transformation for each predetermined quantity of quantized data quantized by the quantizing means; An audio signal compression device is provided, which comprises formatting means for formatting the data compressed by the compression means so as to be arranged in a user data area of a digital disc.

Further, according to the present invention, the audio signal is quantized with 16 bits and the sampling frequency is 44.1 kHz.
a quantizing means for quantizing at a frequency of z or higher; a data compressing means for compressing the data quantity by applying orthogonal transformation for each predetermined quantity of quantized data quantized by the quantizing means; The data compressed by the compression means is CD-R
An audio signal compression apparatus is provided, which has a formatting means for formatting to be arranged in a user data area of mode 2 of the OMXA standard.

Further, according to the present invention, the audio signal is quantized with 16 bits and the sampling frequency is 44.1 kHz.
a quantizing means for quantizing at a frequency of z or higher; a data compressing means for compressing the data quantity by applying orthogonal transformation for each predetermined quantity of quantized data quantized by the quantizing means; An audio signal compression apparatus is provided, which includes formatting means for formatting the data compressed by the compression means so as to be arranged in the user data area of the DVD.

Further, according to the present invention, the audio signal is quantized with 16 bits and the sampling frequency is 44.1 kHz.
Formatting is performed by quantizing at a frequency of z or higher, applying an orthogonal transform to each quantized predetermined amount of quantized data to compress the data amount, and arranging the compressed data in a user data area of a digital disc. Then, an optical recording medium in which the formatted data is recorded in the CD format is provided.

Further, according to the present invention, the audio signal is quantized with 16 bits and the sampling frequency is 44.1 kHz.
Quantize at a frequency of z or higher, apply an orthogonal transform to each quantized predetermined amount of quantized data to compress the data amount, and compress the compressed data in the mode 2 of the CD-ROMXA standard, form 2. There is provided an optical recording medium which is formatted so as to be arranged in the user data area and in which the formatted data is recorded as a CD format.

Further, according to the present invention, the audio signal is quantized with 16 bits and the sampling frequency is 44.1k.
Quantize at a frequency of Hz or higher, apply orthogonal transformation to each quantized predetermined amount of quantized data, compress the data amount, and format the compressed data to be arranged in the user data area of the DVD. Provided is an optical recording medium in which formatted data is recorded in a CD format.

[0019]

BEST MODE FOR CARRYING OUT THE INVENTION Embodiments of an audio signal compression recording apparatus, an audio signal compression apparatus and an optical recording medium according to the present invention will be described with reference to preferred embodiments. FIG. 1 is a block diagram showing a preferred embodiment of an audio signal compression recording apparatus of the present invention. An analog signal such as a music signal is supplied to the input terminal IN, and the output terminal OUT is connected to a CD master making machine (not shown), that is, a mastering device via a premastering device as necessary. Since the mastering device itself is essentially the same as the conventional one, its explanation is omitted here.

The device of FIG. 1 has an A connected to the input terminal IN.
/ D converter 1 and signal processing circuit 2 connected to its output
A memory 3 connected to the signal processing circuit 2 and a CD-ROM encoding circuit 4 connected to the output of the signal processing circuit 2.
And a CD connected to the output of the CD-ROM encoding circuit 4.
It has an encoding circuit 5. CD-ROM encoding circuit 4
Is connected to the first output terminal OUT1 and the output of the CD encoding circuit 5 is connected to the second output terminal OUT2. As described later, the CD encoding circuit 5 may be unnecessary.

The A / D converter 1 operates as a quantizing means for quantizing an audio signal at a quantization bit number of 16 bits and a sampling frequency of 44.1 kHz or higher. The sampling frequency is 44.1 kHz (DV according to the embodiment).
It is either 48 kHz in the case of D) or 88.2 kHz (96 kHz in the case of DVD), but 44.
It can be set to an appropriate value of 1 kHz or more. When targeting music signals, there are usually 2 channels on the left and right, but 4 channels or 6 channels are required depending on the surround and other needs.
It can be a channel or the like. Here, the case of two channels will be described. The quantized data obtained by the A / D converter 1 is written in the memory 3 via the signal processing circuit 2 in units of 2 ^m pieces (m is a positive integer) per channel. Then, the signal processing circuit 2 is the 2 ^m
Start processing of this data.

FIG. 2 is a block diagram showing an example of the signal processing circuit 2. The 2 ^m pieces of data are orthogonally transformed by the orthogonal transformation circuit 10 to obtain a frequency spectrum. A plurality of filters 6 for dividing the frequency spectrum into bands
a, 6b, 6c. . . It is given to the normalization section / quantization section 11 via the filter bank 6 having 6n and the switch circuit 7 as the selection means, and is normalized / quantized collectively for each band. Here, the normalization level (the number of bits) is used as auxiliary information, and the spectrum data is used as main information to form a data frame.
A new data frame is created from this data frame using the codebook index as auxiliary information and the processed data as main information, and this is sequentially written in the memory 3. Next, this new data frame is read from the memory 3 and output to the CD-ROM encoding circuit 4 of FIG. 1 via the allocation circuit 9.

In the CD-ROM encoding circuit 4, a sync signal (SYNC), a header, a subheader, etc. are added to each sector so that a predetermined format described later with reference to FIG. Processing circuit 2
The compressed audio data given by is arranged and output. The output data of the CD-ROM encoding circuit 4 is output via the first output terminal OUT1, recorded on a magnetic tape, for example, and supplied to a premastering device or a mastering device for manufacturing a read-only CD. On the other hand, C
The output data of the D-ROM encoding circuit 4 is a CD in the case of a writable, so-called write-once type CD.
The data is supplied to the encoding circuit 5, is CD-formatted, and is recorded by a recording head (not shown) via the second output terminal OUT2.

Next, some aspects of the present invention will be described with reference to FIG. FIG. 3 shows various formats of the CD in units of sectors. The first stage is a normal music C.
CD-DA, which is D, and various CD-ROMs from the second stage to the sixth stage are shown below. There are the following six modes as examples of the present invention. Note that FIG. 17 showing a DVD
Will be described later.

[0025]

[Table 1] (1) CD-ROM XA mode 2, form 2 (6th stage in FIG. 3) Sampling frequency: 44.1 kHz Quantization bit number: 16 bits (2) CD-ROM XA mode 2, form 2 (6th stage of FIG. 3) Sampling frequency: 88.2 kHz Quantization bit number: 16 bits (3) CD-ROM mode 2 (4th stage of FIG. 3) Sampling frequency: 44.1 kHz Quantization bit number: 16 bits (4) CD-ROM mode 2 (4th stage of FIG. 3) Sampling frequency: 88.2 kHz Quantization bit number: 16 bits (5) DVD (FIG. 17) Sampling frequency: 48 kHz Quantization bit number: 16 bits (6) DVD (Fig. 17) Sampling frequency: 96 kHz Quantization bit number: 16 bits

In the CD-ROM XA mode 2 and form 2, the user data is 2324 bytes. Also, C
In D-ROM mode 2, user data is 2336 bytes. According to these standards, since the data amount of user data, that is, the number of bytes is relatively large, a large amount of data can be recorded and stored in one disk, which is advantageous.

The CD-ROM of the above (1) and (2)
When using XA Mode 2 Form 2, it is possible to specify a subheader with a unique allocation. Table 2 shows the contents of the subheader.

[0028]

[Table 2]

By using bits 5 to 2 of the submode byte in the subheader as the coded ID, it is possible to decode this format while watching the subheader. Tables 3 and 4 below show the submode in the subheader and the content of the coding information. The conditions at the time of formatting can be recorded in the subheader, and there are two methods for that. One of them is a method of inserting the format condition of the sector, and the other is a method of recording the format condition by dividing it into a plurality of sectors. In this case, the information of the plurality of sectors can be collected and decoded. .

[0030]

[Table 3]

[0031]

[Table 4]

In the above four modes, the sampling frequency is 88.
2 kHz, (2) and (4), 1 in 2 blocks
It constitutes a frame. Therefore, 44.1
The time that can be recorded is halved compared to the case of kHz.

In the above embodiment, the case where the signal processing circuit 2 is a reversible compression system has been described. However, by applying the so-called quasi-reversible coding system developed by the present inventors, the data amount can be further increased. Can be compressed. This method will be described below.

FIG. 4 is a block diagram showing an example in which the quasi-reversible coding apparatus for speech developed by the present inventors is applied to the signal processing circuit of FIG. 1, and FIG. 5 is a psychoacoustic analysis and code amount in FIG. FIG. 6 is a flow chart for explaining the adjustment process, FIG. 6 is an explanatory view showing a comparative example of the requantization noise level when the code amount is insufficient in the semi-reversible coding apparatus of FIG. 4 and the conventional example, and FIG. It is explanatory drawing which shows the audio quality comparison example in an encoding device and a prior art example.

In the apparatus shown in FIG. 4, first, similarly to the conventional frequency domain processing encoder, the buffer 21 buffers the PCM signals for the frames necessary for the windowing / orthogonal transformation unit 22 in the subsequent stage to perform the orthogonal transformation. Then, the windowing / orthogonal transformation unit 22 performs windowing (generally, windowing such as Hanning window) on this frame data, performs orthogonal transformation by MDCT (Modified Discrete Cosine Transform) or the like, and outputs the orthogonal transformation coefficients to a plurality of bands. To divide. The normalization unit 23 determines the normalization coefficient (scale factor) for each band and normalizes the orthogonal transform coefficient within the band. The quantizing / encoding unit 24 quantizes the coefficient after the normalization with a precision necessary for reversibility, and performs entropy coding if necessary.

In the example of FIG. 4, the psychoacoustic analysis unit 2
5, the code amount control unit 26, and the quantization / encoding unit 24 are shown in FIG.
The following processing is performed. In FIG. 5, first, the first quantization bit number (Bit [i]) of the coefficient normalized by the quantization / encoding unit 24 is determined, the code amount is estimated, and the total code amount (Total bit) is calculated. Calculate (step S1). Next, the usable code amount (Avail bit) of the frame is confirmed or calculated (step S2), and then the total code amount (Tota) is calculated.
It is checked whether or not the code amount is insufficient by comparing l bit) with the usable code amount (Avail bit) (step S3).

When the code amount is insufficient (Total bi
t> Avail bit), first, the band power p considering the masking effect of the psychoacoustic model and the minimum audible limit characteristic.
A masking curve m [i] is calculated from [i] (= normalized value ² = scale [i] ² ) (step S4). In this case, the masking curve m [i] is obtained by convolving the reference curve curve [i] and the band power p [i].

Next, the standard noise level N [i] of each band is calculated from the minimum audibility limit and the masking curve (step S5), and then the bit is reduced bit by bit from the band having the highest standard noise level N [i]. The insufficient code amount is distributed to each band. However, 6.0 is subtracted from N [i] every time one bit is reduced in band i so that the bit reduction is similar to the standard noise level N [i] (step S6). Then, the quantization / encoding unit 24 requantizes and encodes with the number of quantization bits finally determined for each band in this way (step S7).

If the code amount is not insufficient in step S3, surplus bits are assigned or padded to each band (step S8), and requantization is performed by the quantization / encoding unit 24 with the number of quantization bits. And encoding (step S7). The formatted output unit 27 generally includes the normalization coefficient (the number of quantization bits in some cases),
The code amount control information of the code amount control unit 26 and auxiliary information such as a header are added to the code amount control information to format (bit stream) and transmit.

FIG. 6 shows a comparison between the example of FIG. 4 and an example of setting the requantization noise level when the code amount is insufficient in the conventional encoder. According to the above example, the requantization noise is shaped according to the psychoacoustic model,
Even if the noise amount is the same, the same effect as when the noise level is lowered can be obtained in terms of hearing. Therefore, it is possible to perform quasi-reversible encoding while minimizing the sound quality deterioration in hearing.

FIG. 7 shows a comparative example of the sound quality between the case of lossy encoding in the conventional example and the case of the above example, and FIG. 7A shows the case where a part of the frame is lossy. (B) shows a case where most of the frame is irreversible. In the irreversible section as shown in the figure, the present invention shown by the thick line can improve the sound quality as compared with the conventional example shown by the thin line, and therefore stable sound quality can be obtained as a whole encoding. Further, according to the present invention, it is possible to sufficiently secure the sound quality when the lossy encoding is performed, and therefore, the usable code amount of each frame can be transmitted at a fixed “fixed transmission rate”. , Even if the number of irreversible frames is greatly increased, the sound quality problem does not occur. As a result, the processing relating to the authoring and the code amount control on the reproducing device side can be greatly simplified.

Next, another example of the signal processing circuit is shown in FIG.
It will be described with reference to FIG. Similar to FIG. 4, FIG. 8 is a block diagram showing another example of the quasi-reversible encoding device for speech developed by the present inventors, and FIG. 9 is a flowchart for explaining the process for calculating the code amount correction value in FIG. 10 is a graph showing the relationship between the code amount deviation and the code amount correction value, FIGS.
3 is an explanatory view showing the code amount deviation histogram before and after the code amount correction, FIG. 14 is a flowchart for explaining the psychoacoustic analysis and the code amount adjustment processing in FIG. 8, and FIG. 15 is the semi-reversible encoding in FIG. 16 is an explanatory view showing a comparative example of the requantization noise level when the code amount is excessive between the apparatus and the conventional example, FIG.
FIG. 9 is an explanatory diagram showing a comparative example of auditory sound quality between the semi-reversible encoding device of FIG. 8 and a conventional example.

In the apparatus shown in FIG. 8, first, similar to the conventional frequency domain processing encoder, the buffer 21 buffers the PCM signals for the frames necessary for the windowing / orthogonal transform unit 22 in the subsequent stage to perform the orthogonal transform. Then, the windowing / orthogonal transformation unit 22 performs windowing (generally, windowing such as Hanning window) on the frame data, performs orthogonal transformation by MDCT (Modified Discrete Cosine Transform), etc., and obtains the orthogonal transformation coefficients in a plurality of bands. Split into. The normalization unit 23 determines the normalization coefficient (scale factor) for each band and normalizes the orthogonal transform coefficient within the band. The quantizing / encoding unit 24 quantizes the coefficient after the normalization with an accuracy required for reversibility, and also in this case, performs entropy encoding if necessary. However, FIG.
The effect of entropy coding is generally less than that of the time domain processing shown in FIG.

In this example, the psychoacoustic analysis unit 25
The code amount control unit 26 and the quantization / encoding unit 24 perform the following processing based on the code amount correction value Adj for each section. First, in the present invention, when producing an audio medium, one song (for example, 4 to 6 minutes) or all songs (for example, 40 songs).
(~ 74 minutes) and the like so that the code amount is controlled to a target value on a long-term average, and the encoding process is performed in two passes. Specifically, (a) the first encoding process is performed assuming lossless encoding. However, it is only necessary to obtain the used code amount of each section, and it is not necessary to actually perform quantization / encoding. (B) As shown in FIG. 9, the code amount correction value Adj of each section is calculated from the difference between the used code amount of each section and the target code amount. (C) Perform the second encoding process. In this case, bit allocation assuming reversible coding is changed by the correction code amount and the psychoacoustic model to perform quantization / coding, and
The bit allocation change information is transmitted to the decoder as auxiliary information.

Next, the process for calculating the code amount correction value Adj in (b) above will be described with reference to FIG. First, the average code amount T is input by inputting the used code amount of the target section.
m is calculated, and the difference from the target code amount Td is evaluated (steps S11 and S12). Next, when the code amount is excessive (average code amount Tm> target code amount Td), the deviation Delta [bit] between the used code amount and the target code amount of each section is calculated (however, positive when excess). , The deviation Delta [bit] is quantized with an appropriate step width step [bit] to create a histogram (steps S12 → S1).
3). Next, the total deviation S in the area where the deviation of the histogram is negative
m and the total deviation Sp of the positive region are calculated as follows (step S14).

[0046]

(Equation 1)

Next, when the ratio Sm / (Sm + Sp) of the total deviation Sm in the negative region is larger than a predetermined value Bound (for example, 0.33), the code amount correction value for each section is as follows. Find Adj (step S15 → S1)
6).

[0048]

(Equation 2)

On the other hand, the ratio Sm / (Sm + Sp) is a value
If smaller than Bound, the ratio Sm / (Sm + Sp)
Is determined to be larger than the value Bound (step S15 → S17), and the code amount correction value Adj for each section is calculated as follows (step S18).

[0050]

(Equation 3) Here, the reason for using this method is that when the histogram is extremely biased toward the "excessive" side, it is necessary to apply an offset to all frames to correct it.

If the average code amount Tm> the target code amount Td is not satisfied in step S12, a constant code amount correction value Adj in each section is set as follows based on the average code amount Tm and the target code amount Td. Obtained (step S19). Adj = (Td-Tm) [bit]

FIG. 10 shows the relationship between the code amount deviation Delta [bit] and the code amount correction value Adj. As the deviation Delta [bit] is positive and larger, the correction value Adj also increases. Also, FIGS.
Shows histograms before correction (solid line) and after correction (broken line), the horizontal axis shows deviation per sample (Delta / number of samples per section), and the vertical axis shows frequency. Specifically, FIG. 11 shows the case where the correction value Adj is obtained as described above, and FIG. 12 and FIG.
The case where the correction value Adj is obtained is shown as.

Next, the psychoacoustic analysis and the code amount adjustment processing will be described with reference to FIG. In FIG. 14, first, the number of quantization bits (Bit [i]) for the first time (reversible method) of the coefficients normalized by the quantization / encoding unit 24 is determined, and the total amount of code ( Total bit) is calculated (step S21). Next, the code amount correction value Adj for that frame
(Step S22), the correction value Adj is negative (Adj
It is checked whether or not <0) (step S23).

When the correction value Adj is negative (code amount reduction
First, the masking effect of the psychoacoustic model and the maximum
Band power p [i] (= normalization considering small audible limit characteristics
coefficient 2 ＝ scale [i]^Two ) Calculate masking curve m [i]
Take out (step S4). In this case, the masking curve
m [i] is the standard curve curve [i] and band power p [i]
It is obtained by performing a calculation.

Next, the standard noise level N [i] of each band is calculated from the minimum audibility limit and the masking curve (step S5), and then bit reduction is performed bit by bit from the band having the highest standard noise level N [i]. The code amount correction value is assigned to each band. However, 1 in band i
Each time bit reduction is performed, 6.0 is subtracted from N [i] so that the bit reduction becomes similar to the standard noise level N [i] (step S6). Then, the quantization / encoding unit 24 requantizes and encodes with the number of quantization bits finally determined for each band in this way (step S7).

Further, in step S23, the correction value Adj
If is not negative (increased code amount), surplus bits are assigned or padded to each band (step S8), and the quantization / encoding unit 24 requantizes and encodes with the number of quantization bits ( Step S7). The formatting output unit 27 generally adds a normalization coefficient (the number of quantization bits in some cases), the code amount control information of the code amount control unit 26, and auxiliary information such as a header to the formatting (bitstream conversion). ) And transmit.

Therefore, according to the above example, the interval where the arithmetic entropy is large and the auditory entropy is small,
Since more code amount correction (reduction) is performed, the code amount can be distributed according to the sense of hearing. Further, FIG. 15 shows a case where the apparatus of FIG. 8 is compared with an example of setting the requantization noise level when the code amount is excessive in the conventional encoder. According to this example, even in a frame to be lossy encoded. Requantization noise is shaped according to the psychoacoustic model, and even if the noise amount is the same, it is possible to obtain the same effect as when the noise level is lowered in the sense of hearing. Therefore, it is possible to perform quasi-reversible encoding while minimizing the sound quality deterioration in hearing.

FIG. 16 shows a comparative example of the sound quality between the case where lossy encoding is performed in the conventional example and the case of the example of FIG. 8 above. FIG. 16A shows the case where a part of the frame is lossy. 16 (b) shows a case where most of the frame is irreversible. In the irreversible section as shown in the figure, the present invention shown by the thick line can improve the sound quality as compared with the conventional example shown by the thin line, and therefore stable sound quality can be obtained as a whole encoding.

Next, the data compression rate according to the present invention will be examined. Table 5 shows the results of actually measuring the compression effect for each of the three music genres.
Note that the first stage 1644 in the table indicates that the number of bits is 16 and the sampling frequency is 44.1 kHzz. In each genre, 5 to 10 songs were selected and surveyed.

[0060]

[Table 5]

The numbers in Table 5 indicate the data amount after compression when the base data amount is 100. As can be seen from this table, for example, when 1644 is applied to classical music, it is possible to compress 50% on average and 60% at maximum. It's not possible to compress as much as classical music in pops and jazz fusion, but on average 23% to 35%
% Compression ratio is obtained.

The effect of compression shown in Table 5 is shown in FIGS.
This is a case where the quasi-lossless coding device shown in is not used, and the compression rate can be further increased by using the quasi-lossless coding device.

According to the four modes of the embodiment of the present invention described above, a user of the compression recording apparatus or the compression apparatus of the present invention manually operates a select button (not shown) or the like so that any one of them can be selected. , Can be switched and used. The sampling frequency is 44.1k.
When it is set higher than Hz, it is necessary to control the rotational speed of the disk so that the linear velocity becomes higher than the constant linear velocity at 44.1 kHz. Sampling frequency is 44.1
When the frequency is set higher than kHz, the high frequency characteristics are improved and the sound quality is improved. FIG. 17 is a schematic data layout diagram showing the format of a DVD in sector units as in FIG. As shown in FIG. 17, it is usually 1 for DVD.
The pack consists of 2048 bytes (1 logical sector),
2034 bytes of the packet (user data) in it can be used. In addition, in FIG. 17, "pack start"
Has a SYNC pattern that serves as a synchronization signal, and "SCR"
Is a system clock reference which is time information, "Mux rate" is a transfer rate (multiplexing rate), and "packet (user data)" is composed of a packet header and data.

[0064]

As described above, according to the present invention, an audio signal is quantized with a quantization bit number of 16 bits and a sampling frequency of 44.1 kHz or more, and a predetermined amount of quantized quantization is performed. Apply orthogonal transformation to each data to compress the data amount and compress the compressed data to CD-ROM
The format or the user data area of the CD-ROM standard or the user data area of the CD-ROM standard or the DVD or the user data area of the mode 2 of the XA standard or form 2 or the CD-ROM standard. Since it is formatted to be placed in
Audio signals are compressed at a high compression rate, CD-ROM or DVD
Can be recorded. Therefore, depending on the content of the music, the recording time can be longer than that of the CD-DA. In addition, the new standard DVD audio can be realized.

[Brief description of the drawings]

FIG. 1 is a block diagram showing a preferred embodiment of an audio signal compression recording apparatus of the present invention.

FIG. 2 is a block diagram showing an example of a signal processing circuit 2 in FIG.

FIG. 3 is a schematic diagram of data arrangement showing various formats of a CD in units of sectors.

FIG. 4 is a block diagram showing an example of a quasi-lossless coding apparatus for speech as another example of the signal processing circuit 2 in FIG.

5 is a flowchart for explaining the psychoacoustic analysis and code amount adjustment processing in FIG.

FIG. 6 is an explanatory diagram showing a comparative example of requantization noise levels when the code amount is insufficient in the semi-reversible encoding device of FIG. 4 and a conventional example.

FIG. 7 is an explanatory diagram showing a comparative example of auditory sound quality between the semi-reversible encoding device of FIG. 4 and a conventional example.

8 is a block diagram showing an example of a quasi-reversible coding apparatus for speech as still another example of the signal processing circuit 2 in FIG.

9 is a flowchart for explaining a process of calculating a code amount correction value in FIG.

FIG. 10 is a graph showing the relationship between code amount deviation and code amount correction value.

FIG. 11 is an explanatory diagram showing histograms of code amount deviations before and after code amount correction.

FIG. 12 is an explanatory diagram showing histograms of code amount deviations before and after code amount correction.

FIG. 13 is an explanatory diagram showing code amount deviation histograms before and after code amount correction.

FIG. 14 is a flowchart for explaining the psychoacoustic analysis and code amount adjustment processing in FIG.

FIG. 15 is an explanatory diagram showing a comparative example of the requantization noise level when the code amount is insufficient in the semi-reversible encoding device of FIG. 8 and the conventional example.

16 is an explanatory diagram showing an audio quality comparison example between the semi-reversible encoding apparatus of FIG. 8 and a conventional example.

FIG. 17 is a schematic diagram of data arrangement showing a DVD format in sector units.

[Explanation of symbols]

DESCRIPTION OF SYMBOLS 1 A / D conversion circuit (quantization means) 2 Signal processing part (it comprises a data compression means with the memory 3) 3 Memory 4 CD-ROM encoding circuit (formatting means) 5 CD encoding circuit 10 Orthogonal conversion circuit 11 Normal Conversion unit 22 Windowing / Orthogonal transformation unit 23 Normalization unit 24 Quantization / coding unit 25 Auditory psychological analysis unit 26 Code amount control unit 27 Formatting output unit IN input terminals OUT1 and OUT2 output terminals

Claims (13)

[Claims] 1. Quantizing means for quantizing an audio signal at a quantization bit number of 16 bits and a sampling frequency of 44.1 kHz or higher, and a predetermined amount of quantized data quantized by the quantizing means. Data compression means for applying orthogonal transformation for each data to compress the data amount, formatting means for formatting the data compressed by the data compression means so as to be arranged in the user data area of the digital disc, and formatting by the formatting means. And a means for recording the recorded data as a CD format on a recording medium. 2. Quantizing means for quantizing an audio signal with a quantization bit number of 16 bits and a sampling frequency of 44.1 kHz or higher, and a predetermined amount of quantized data quantized by the quantizing means. An audio signal compression apparatus comprising: a data compression unit that applies an orthogonal transformation for each data to compress a data amount; and a formatting unit that formats the data compressed by the data compression unit so as to be arranged in a user data area of a digital disc. . 3. An audio signal is quantized with a quantization bit number of 16 bits and a sampling frequency of 44.1 kHz or higher, and an orthogonal transform is applied to each quantized predetermined amount of quantized data to obtain a data amount. An optical recording medium in which the compressed data is formatted, the compressed data is formatted so as to be arranged in a user data area of a digital disc, and the formatted data is recorded in a CD format. 4. Quantizing means for quantizing an audio signal at a quantization bit number of 16 bits and a sampling frequency of 44.1 kHz or higher, and a predetermined amount of quantized data quantized by the quantizing means. A data compression unit for compressing the data amount by applying orthogonal transformation for each of the data; and a CD-ROM for storing the data compressed by the data compression unit.
XA standard mode 2, form 2, or CD-RO
An audio signal compression recording apparatus comprising: formatting means for formatting so as to be arranged in a user data area of M standard mode 2; and means for recording the data formatted by the formatting means in a recording medium as a CD format. 5. Quantizing means for quantizing an audio signal at a quantization bit number of 16 bits and a sampling frequency of 44.1 kHz or higher, and a predetermined amount of quantized data quantized by the quantizing means. A data compression unit for compressing the data amount by applying orthogonal transformation for each of the data; and a CD-ROM for storing the data compressed by the data compression unit.
XA standard mode 2, form 2, or CD-RO
An audio signal compression apparatus, comprising: formatting means for formatting so as to be arranged in a user data area of mode 2 of M standard. 6. An audio signal is quantized with a quantization bit number of 16 bits and a sampling frequency of 44.1 kHz or higher, and orthogonal transform is applied to each quantized predetermined amount of quantized data to obtain a data amount. A compressed optical disk, and the compressed data is formatted so as to be arranged in the user data area of the CD-ROM XA standard mode 2, the form 2 or the CD-ROM standard mode 2, and the formatted data is recorded in the CD format. recoding media. 7. Quantizing means for quantizing an audio signal at a quantization bit number of 16 bits and a sampling frequency of 44.1 kHz or higher, and a predetermined amount of quantized data quantized by the quantizing means. A data compression unit that applies orthogonal transformation for each data to compress the data amount, a formatting unit that formats the data compressed by the data compression unit so as to be arranged in the user data area of the DVD, and a format that is formatted by the formatting unit. A device for recording data in a recording medium as a CD format, the audio signal compression recording device. 8. The audio signal is a two-channel signal, and the data compression means is 2 ^m per channel.
The audio signal compression recording according to any one of claims 1, 4, and 7, wherein the orthogonal transformation is applied to each piece (m is a positive integer) of quantized data to compress the data amount. apparatus. 9. A means for the data compressing means to frame the audio signal for each predetermined section length, and a code amount required to encode the signal in the frame by a reversible method, A code amount control means for comparing with the usable code amount, a psychoacoustic analysis means for analyzing the signal in the frame by a psychoacoustic model, and a signal in the frame is reversible when the frame code amount is less than or equal to the usable code amount. Irreversible quantizing means for quantizing the signal in the frame by an irreversible method based on the output of the psychoacoustic analyzing means when the frame code quantity exceeds the usable code quantity. Item 9. The audio signal compression recording device according to any one of items 1, 4, 7, and 8. 10. A framing means for framing the audio signal for each predetermined section length by the data compression means, and calculating a difference between a target code amount and an actual code amount by a reversible method for all sections to be encoded. , A code amount correction value calculation means for calculating a correction value according to the excess or deficiency of the code amount for each section, a psychoacoustic analysis means for analyzing a signal in a frame with a psychoacoustic model, and an average code quantity for all sections Irreversible quantization in which the signal in each section is quantized by a reversible method based on the code amount correction value so that the target code amount becomes, or irreversibly quantized based on the output of the psychoacoustic analysis means. An audio signal compression recording apparatus according to any one of claims 1, 4, 7, and 8, further comprising: 11. The data compression means comprises orthogonal transformation means for orthogonally transforming the quantized signal, band division means for dividing the data orthogonally transformed by the orthogonal transformation means into a plurality of bands, and the band. 11. The normalizing means capable of responding to the output signal of the dividing means, and the selecting means for supplying the band-divided data by the band dividing means to the normalizing means for each band.
The audio signal compression recording device according to any one of 1. 12. The number of quantization bits of an audio signal is 16
Bits, a quantizing means for quantizing at a sampling frequency of 44.1 kHz or higher, and orthogonal transformation for each predetermined quantity of quantized data quantized by the quantizing means to compress the data quantity. An audio signal compression apparatus comprising: a data compression unit; and a formatting unit that formats the data compressed by the data compression unit so as to be arranged in a user data area of a DVD. 13. An audio signal having a quantization bit number of 16
Bit, sampling frequency is quantized at a frequency of 44.1 kHz or higher, orthogonal transformation is applied to each quantized predetermined amount of quantized data to compress the data amount, and the compressed data is used as DVD user data. Format the data so that it will be placed in the area, and put the formatted data on a CD.
An optical recording medium recorded as a format.