JP2810030B2 - Data processing method and signal processing device - Google Patents

Description

DETAILED DESCRIPTION OF THE INVENTION

[0001]

BACKGROUND OF THE INVENTION 1. Field of the Invention The present invention relates to a data arrangement method and a medium for recording or transmitting data recorded on a digital video disk or a digital audio disk, and a signal processing apparatus therefor.

[0002]

2. Description of the Related Art Recently, a digital video disk and a reproducing apparatus thereof have been developed as an optical disk in addition to a conventional audio compact disk (hereinafter, referred to as a CD). In this digital video disc,
In particular, recently, a disc which is about the same size as a conventional CD (12 cm in diameter) and capable of recording and reproducing movie information for about 2 hours has been developed. In addition, in this digital video disc, a format is considered in which, in addition to movie information, audio or music in eight different languages and subtitle information in 32 different languages can be recorded on the same disc.

[0003]

As described above, in recent years, digital video discs capable of recording voices or music in various languages in addition to main movie information and having the same size as a conventional CD have been developed. Is being developed.

[0004] When such digital video discs are available on the market, there is naturally a demand to play music and voice (audio signals) of a conventional CD and a new digital video disc by an audio-only player. come. There are a compression method and a linear PCM method as an audio signal recording method. When a video disk capable of reproducing music and audio signals in an audio-only player is considered, data is recorded in a linear PCM method similar to a conventional CD. It is effective to record Also, in the case of video disc players, there is a high possibility that simple models and high-quality models will be on the market.

Therefore, according to the present invention, other channel signals of higher quality than conventional CDs can be recorded, and a simple model can be used.
An object of the present invention is to provide a data arrangement method and a medium for recording or transmitting data, which are effective for recording or processing linear PCM data or the like, which can be easily reproduced by any of higher-order models, and a processing apparatus therefor.

[0006]

According to the present invention, in order to achieve the above object, one-channel or plural-channel signals are sampled in time series, and quantized data is recorded or transmitted for reproduction. Data for each channel is sampled from the M-bit main word on the MSB side and the LSB.
And the main word of the (2n) -th sample data of each channel is collectively arranged as a main sample S2n. Then, the main word of the (2n + 1) -th sample data of each channel is arranged. The words are collectively arranged as a main sample S2n + 1, then the extra words of the (2n) -th sample data of each channel are arranged and arranged as an extra sample e2n, and then the (2n +
1) The extra words of the sample data are collectively arranged as an extra sample e2n + 1 (where n =
0, 1, 2,...).

With this configuration, the reproduction processing circuit is simplified in a simple model that reproduces only the main word or only the main word for two channels, and an extra word reproduction circuit is added to the main word reproduction circuit in the higher model. It is good to do.

[0008]

Embodiments of the present invention will be described below with reference to the drawings.

First, in the data recording method of the present invention,
An arrangement of data according to the linear PCM method will be described. For the linear PCM data, for example, 16 bits, 20 bits, and 24 bits are arbitrarily adopted as quantization bits. Further, audio modes include monaural, stereo, three-channel, four-channel, five-channel, six-channel, seven-channel, and eight-channel modes.

Now, it is assumed that there are audio signals of eight channels A to H. These are 48 kHz or 9
It is sampled at a sampling frequency of 6 KHz and quantized. The quantization bit will be described by taking 20 bits as an example.

FIG. 1A shows a state in which audio signals A to H of up to eight channels are sampled, respectively. It is assumed that each sample is quantized to, for example, 20 bits. 2 more
Each sample of 0 bits indicates a state of being divided into a main word and an extra word.

The main word of each channel is indicated by a capital letter An-Hn, and the extra word is indicated by a small letter an-hn. The suffix n (n
= 0, 1, 2, 3, 3,...) Indicates a sample order. Here, the main word is 16 bits, and the extra word is 4 bits.

The signal A is A0 a0, A1 a1, A2 a2
, A3 a3, A4 a4..., The signal B is B0 b0
, B1 b1, B2 b2, B3 b3, B4 b4..., The signal C is represented by C0 c0, C1 c1, C2 c2, C3 c
The signal H is represented by H0 h0, H1 h1 like 3, C4 c4.
, H2 h2, H3 h3, H4 h4...

Next, FIG. 1B shows the arrangement format of the above words in a sample sequence when the above words are recorded on a recording medium.

First, each sample data of 20 (= M) bits is divided into a main word of 16 (= m1) bits on the MSB side and an extra word of 4 (= m2) bits on the LSB side. Next, 0 (= 2
The n) th main word is arranged collectively. Next to this, the first (= 2n + 1) -th main word of each channel is arranged collectively. Next, the 0th (= 2n) th extra word of each channel is arranged collectively. Then, 1 (= 2n + 1) of each channel
The extra words are grouped together (where n
= 0, 1, 2,...).

Here, a group in which the main words of each channel are collected is defined as one main sample. A group of extra words of each channel is defined as one extra sample.

Due to such a format, when performing data reproduction processing with a simple model (for example, a model operating in a 16-bit mode), it is sufficient to handle only the main word and perform the reproduction processing. When performing data reproduction processing using a model operating in the 20-bit mode), the reproduction processing may be performed by handling the main word and the corresponding extra word.

FIG. 1C shows the arrangement state of each sample using the specific bit numbers of the main sample and the extra sample.

As described above, in the state of the quantized linear PCM code, the following can be achieved by dividing a 20-bit code into a 16-bit main word and a 4-bit extra word. It is. Models that operate in 16-bit mode have
In the extra sample area, unnecessary data can be easily discarded by performing data processing in units of 8 bits. This is because two extra samples are 4 bits × 8 channels and 4 bits × 8 channels. This is because this data can be processed (discarded) continuously eight times in 8-bit units.

The features of this data array are not limited to this embodiment. In both cases where the number of channels is odd and the number of extra words is 8 bits, the total number of bits of two consecutive extra samples is an integral multiple of 8 bits. By executing the 8-bit consecutive discarding process n times according to, the extra sample can be skipped.

In the state shown in FIG. 1 (B), after that, modulation processing may be performed and recorded on a recording medium. However, when recording together with other control information and video information, data handling and In order to facilitate synchronization, it is preferable to record in a form that allows easy time management. Therefore, the following framing, frame grouping, and packetization are performed.

FIG. 1D shows an audio frame sequence. That is, first, the unit of data during the fixed reproduction time is (1/600 second), and this is defined as one frame.

In one frame, 80 or 160 samples are allocated. 48K sampling frequency
In Hz, one sample is 1/4800 seconds,
(1/48000) × 80 samples = 1/600 seconds. When the sampling frequency is 96 KHz, 1
The sample is 1/9600 seconds and (1/9600
0) × 160 samples = 1/600 second. Thus, one frame is made up of 80 samples or 160 samples.

FIG. 2 shows one frame and one GOF.
(Group of frames). One frame is data of 80 or 160 samples and is 1/600 second, and one GOF is composed of 20 frames. Then, this 1 GOF is (1/600) seconds × 20 = 1/30 seconds. That is, this is the frequency of one frame of the television. Such a sequence of GOFs is an audio stream. The unit of one GOF is effective when synchronizing with a video signal. In addition, the above frame
For convenience of recording on the same recording medium as other control signals and video signals, they are allocated to packets. The relationship between the packet and the frame will be described below.

FIG. 3A shows the relationship between the packet and the frame.

DSI is data search information, V is a video object, A is an audio object, S is a sub-picture object, and each block is called a pack. 1 pack is 20
It is specified as 48 bytes. One pack includes one packet, and one pack includes a pack header, a packet header, and a packet. In the DSI, information for controlling each data at the time of reproduction, such as a start address and an end address of each pack, is described.

FIG. 3B shows only the audio pack taken out. Actually, as shown in FIG. 3A, the DSI pack, the video pack V, and the audio pack A are arranged in a mixed manner, but FIG. 3B shows the relationship between the frame and the pack in an easy-to-understand manner. In this case, the audio pack A is extracted and shown in FIG. The standard of this system stipulates that information that is only about 0.5 second when reproducing between the DSI and the next DSI is arranged. Accordingly, since one frame is 1/600 second as described above, the number of audio frames between DSI is 30 frames. The data amount (D) of one frame differs depending on the sampling frequency (fs), the number of channels (N), and the number of quantization bits (m).

When fs = 48 kHz, D = 80 × N ×
When m and fs = 96 kHz, D = 160 × N × m.

Therefore, one frame does not always correspond to one pack, and a plurality of frames may correspond to one pack, or one frame or less may correspond to one pack. That is, as shown in FIG. 3B, the beginning of the frame may come in the middle of one pack. The position information of the head of the frame is described in the pack header, and is described as a data count (timing) from the pack header or DSI. Therefore, the playback device
When reproducing the above-described recording medium, a frame of an audio packet is extracted, and data of a channel to be reproduced is extracted, taken into an audio decoder, and subjected to decoding processing.

FIG. 4A shows the relationship between the main word (16 bits) and the extra word (4 bits) in the 20-bit mode, which generally shows the above data arrangement. FIG. The relationship between the main word (16 bits) and the extra word (8 bits) in the 24-bit mode is shown.

As shown in FIGS. 4A and 4B, the sample data is composed of a main sample and an extra sample as a pair, and two pairs of samples as one unit. Done.

In the above-described signal format, it has been described that interleave processing is not performed.
However, in the case where the recording medium is damaged or data is continuously lost during data transmission, by performing interleaving, continuous loss of signals can be reduced. It is well known that this allows approximate interpolation of missing sample data.

FIG. 5 shows the principle of interleaving and de-interleaving for the above-mentioned format. According to the data arrangement of the present invention, even when interleaving is performed, deinterleaving of only the main word is easy in a simple model, and it is advantageous that the circuit can be simplified.

This example is a delay interleave method in which the interleave length D is 2 k samples. Here, S means one main sample, and main sample S0 = A0, B0,
.. H0, S1 = A1, B1,... H1, S2 = A2,
B2,... H2, and Sj = Aj, Bj, Cj,.

E means an extra sample, and extra samples e0 = a0, b0,... H0, e1 = a
, H1, e2 = a2, b2, ... h2, and ej = aj, bj, cj, .... The even sample of the main sample is input to the transmission system L11 without delay,
Odd samples of the main sample are input to the delay transmission system L12. The even sample of the extra sample is
The odd-numbered extra samples are input to the delay transmission system L14.

Regarding the delay amount of the extra sample,
When the extra sample is 4 bits, the amount of delay may be 1/4 of the delay amount of the main sample (16 bits). When the extra sample is 8 bits, the delay amount may be 1 compared to the delay amount of the main sample (16 bits). / 2 is sufficient. Therefore, the delay transmission system L14 can switch the delay amount between the 20-bit mode and the 24-bit mode.

Here, looking at the vertical column of each sample on the input side of the transmission system in FIG. 5, the format described in FIG. 1B is maintained. Thus, the columns are synchronized and each sample is input to the corresponding transmission system. Then, a two-dimensional array of samples as shown on the right side of each transmission system is obtained. Looking at the vertical columns of this two-dimensional array, although the data content is different from that before interleaving, it is possible to obtain an array of a combination of two main samples and two extra samples described in FIG. 1B.

When performing deinterleaving, an even-numbered sample sequence of main samples is input to a delay transmission line, and an odd-numbered sample sequence is input to a delay-free transmission line. Further, an even-numbered sample sequence of the extra samples is input to the delay transmission line, and an odd-numbered sample sequence is input to the transmission line without delay. By such processing, the original sample sequence can be obtained. In the case of the 16-bit mode, only the transmission system of the main sample needs to be used.

On the reproduction side, a deinterleave circuit using only main samples may be used in a model that reproduces only main samples. Further, when only a specific channel is reproduced, a deinterleave circuit for a word of only the specific channel in the sample data is used.

As described above, it is possible to obtain an arrangement method, a medium, and a processing apparatus for recording or transmitting data of a multi-channel linear PCM system capable of performing reproduction processing in both a simple model and a high-order model. Can be.

FIG. 6A shows an example of the arrangement of packs including packets.

DSI is data search information, V is a video object, A is an audio object, S is a sub-picture object, and each block is called a pack. 1 pack is 2
048 bytes, which is fixed. One pack is
One pack includes one packet, and one pack includes a pack header, a packet header, and a packet data part. In the DSI, information for controlling each data at the time of reproduction, such as a start address and an end address of each pack, is described.

FIG. 6B shows only the audio pack taken out. Actually, a DSI pack, a video pack, and an audio pack are mixedly arranged as shown in FIG. 6A, but in FIG. Shown out. The standard of this system stipulates that an information amount of about 0.5 seconds is arranged when information between the DSI and the next DSI is reproduced. One pack includes a pack header, a packet header, and a packet data part.

Here, the pack header and the packet header include the size of the audio pack, the presentation time stamp for obtaining the reproduction output timing with the video, the identification code of the channel (stream), the quantization bit, the sampling frequency, and the data of the data. Information necessary for reproducing audio, such as a start address and an end address, is described.

Next, the audio inserted in the packet is inserted in units of two pairs of two main samples and two extra samples shown in FIGS. 1 (A) to 1 (C).

FIG. 7 shows an enlarged audio pack. In the data portion of this audio pack, the head of the pair of samples (A0-H0,
A1-H1), and are subsequently arranged in units of two pairs. Here, the number of bytes in one pack is fixed at 2048 bytes. On the other hand, since the sample is variable-length data, 2048 bytes is not necessarily a byte length that is an integral multiple of two pairs of samples. Therefore, the maximum byte length of one pack may be different from the byte length of (2 pairs of samples × an integer). In such a case, the byte length of the pack should be equal to or larger than (2 pairs of samples × an integer multiple). If a part of the pack is left, the remaining part is 7 bytes or less. A stuffing byte is inserted in the header, and if it exceeds 7 bytes, a padding packet is inserted at the end of the pack.

In the case of such pack-format audio information, handling is easy during reproduction.

Since the audio data at the head of each pack is always the head of two pairs of samples, that is, the main sample, the reproduction process is facilitated when the reproduction process is performed at a proper timing. This is because the playback device fetches data in packs and performs data processing. Assuming that audio data samples are arranged across two packs, the two packs are taken in, the audio data is integrated and decoded, and the processing becomes complicated. However, as in the present invention, the audio data at the head of each pack is always the head of two pairs of main samples, and if audio data is grouped in packs, the timing is taken only for one pack. And the processing is easy. Further, since the data processing is performed in units of packets, the authoring system (support system) can be simplified, and the software for data processing can be simplified.

In particular, at the time of special reproduction or the like, video data may be intermittently thinned out for processing or interpolated for processing. In such a case, audio data may be handled in packet units. As a result, it is possible to relatively easily control the reproduction timing. It does not complicate the decoder software.

In the above system, the sample is created in such a manner that the sample is divided into upper 16 bits and lower 4 bits, but the data need not always be in such a format. What is necessary is just to sample linear PCM audio data.

For example, if the data length of the extra sample is 0
Considering the above, the data sequence is a sequence of main samples, and has a general data format. In this case, since there is no extra sample, it is not necessary to use two pairs of samples, and packetization may be performed in units of main samples.

FIG. 8 shows a list of the sizes of linear PCM data when linear PCM data is arranged in a packet in units of two pairs as described above. The number of monaural, stereo, and multi-channel sections is shown, and in each section, the maximum number of samples that can be accommodated in one packet is shown for each quantization bit number. Since the number of samples is two, the number of samples in one packet is all even. As the number of channels increases, the number of bytes increases accordingly, so the number of samples in one packet decreases. When the number of quantization bits is 16 bits and monaural, the number of samples in one packet is 1004, the number of bytes is 2008, the stuffing byte is 5 bytes, and there is no padding byte. However, the stuffing byte of the first packet is 2 bytes. This is because 3-byte attribute information may be added to the header of the first packet.

Looking at the 24-bit quantization bits and the stereo mode, it is shown that the leading packet is stuffed with 6 bytes and the subsequent packets are padded with 9 bytes.

FIG. 9 shows the procedure of the apparatus for generating a pack.

For example, it is assumed that audio signals of each channel are sampled to generate samples as shown in FIG. 1B and stored in a memory. Step S
At 11, samples are taken in order. Step S
At 12, it is determined whether or not the number of bytes has reached the capacity of the packet (2010 bytes). If the number of bytes has reached 2010 bytes, the packet is packed with the sample (step S13).

If the number of bytes is not the capacity of the packet (2010 bytes), the process proceeds to step S14. In step S14, the number of bytes of the acquired sample is 20
Determine if it exceeds 10 bytes. If not, the process returns to step S11. If so, in step S15, the last sample taken is returned to the position of step S11, and the difference between the remaining number of bytes and 2010 bytes is calculated. Here, it is determined whether the difference R exceeds 8 bytes (step S1).
6). When the difference R exceeds 8 bytes, a packet is formed by padding processing (step S17), and when the difference R is 7 bytes or less, a packet is formed by stuffing processing (step S18).

Next, a brief description will be given of a reproducing apparatus for reproducing the above data.

FIG. 10 shows an optical disk reproducing apparatus, FIG. 11 shows a basic configuration of a disk drive section 30 for driving the optical disk 10 on which the above-described audio stream is recorded, and FIG. FIG. 4 shows a diagram for explaining an example.

The optical disk reproducing apparatus shown in FIG. 10 will be described.

The optical disk reproducing device has a key operation / display unit 500. The optical disc playback device has a monitor 1
1. The speaker 12 is connected. The pickup data read from the optical disk 10 is sent to the system processing unit 504 via the disk drive unit 501. The pickup data read from the optical disc 10 is
For example, it includes video data, sub-video data, and audio data, and these data are separated by the system processing unit 504. The separated video data is stored in a video buffer 506.
Is supplied to the video decoder 508 via the sub-picture buffer 507, and the sub-picture data is supplied to the
9, the audio data is supplied to an audio buffer 512.
Is supplied to the audio decoder 513 via the. The video signal decoded by the video decoder 508 and the sub video signal decoded by the sub video decoder 509 are synthesized by the synthesizing unit 510, output as an analog video signal by the D / A converter 511, and supplied to the monitor 11. . The audio signal decoded by the audio decoder 513 is
The analog audio signal is supplied to the speaker 12 by the D / A converter 514.

Reference numeral 502 denotes a system CPU, and the entire reproduction apparatus is managed by the system CPU 502. Therefore, the system CPU 502 can exchange control signals, timing signals, and the like with the disk drive unit 501, the system processing unit 504, and the key operation / display unit 500. A system ROM / RAM 503 is connected to the system CPU 502. The system ROM / RAM 503 stores a fixed program for the system CPU 502 to perform data processing, and stores management data and the like reproduced from the optical disk 10. It can also be stored.

The data RAM 505 is provided in the system processing unit 5
04, and is used as a buffer when performing the above-described data separation, error correction, and the like.

The disk drive unit 501 shown in FIG. 11 will be described.

The disk motor drive circuit 531 drives the spindle motor 532 to rotate. Spindle motor 5
When 32 rotates, the optical disk 10 rotates, and the optical head unit 533 can pick up recorded data recorded on the optical disk. Optical head 5
The signal read by 33 is supplied to the head amplifier 534, and the output of the head amplifier 534 is input to the system processing unit 504.

The feed motor 535 is driven by a feed motor drive circuit 536. Feed motor 53
5 drives the optical head unit 533 in the radial direction of the optical disk 10. The optical head unit 533 is provided with a focus mechanism and a tracking mechanism, and these mechanisms include a focus circuit 537 and a tracking circuit 53, respectively.
8 is provided.

Control signals are input from the servo processing unit 539 to the disk motor drive circuit 531, feed motor drive circuit 536, focus circuit 537, and tracking circuit 538. Accordingly, the disk motor 532 controls the rotation of the optical disk 10 so that the frequency of the pickup signal is a predetermined frequency, and the focus circuit 537 focuses the optical beam of the optical head unit 533 on the optical disk 10 at the best focus. Control the focusing mechanism of the optical system,
Controls the tracking mechanism so that the optical beam is applied to the center of a desired recording track.

The structure of the optical disk 10 shown in FIG. 12 will be described.

The optical disk 10 has an information recording area 22 around the clamp area 21 on both sides. The information recording area 22 has a lead-out area 23 where no information is recorded on the outer circumference, and a lead-in area 24 where no information is recorded at a boundary with the clamp area 21. A space between the lead-out area 23 and the lead-in area 24 is a data recording area 25.

In the data recording area 25, tracks are continuously formed in a spiral shape. This track is divided into a plurality of physical sectors, and the sectors are numbered consecutively. The signal trace of the track is formed as a pit. In a read-only optical disk, a pit row is formed on a transparent substrate by a stamper, and a reflection film is formed on the pit row forming surface to form a recording layer. An optical disc of the two-sheet bonding type is a composite disc in which two discs are combined via an adhesive layer so that such recording layers face each other.

Next, the logical format of the optical disk 10 will be described.

FIG. 13 shows a logical format which is an information division of the information recording area 25. This logical format conforms to certain standards, such as micro-UDF and IS
It is determined in accordance with O9660. In the following description, the logical address is the micro UDF and ISO966
It means a logical sector number (LSN) defined by 0, and the logical sector has the same size as the previous physical sector, and one logical sector is 2048 bytes. Further, it is assumed that the logical sector numbers (LSN) are sequentially numbered together with the ascending order of the physical sector numbers.

The logical format has a hierarchical structure, and has a volume and file structure area 70, a video manager jar 71, at least one or more video title sets 72, and another recording area 73. These areas are partitioned on logical sector boundaries. One logical sector is 2048 bytes. One logical block is also 204
It is 8 bytes, and one logical sector is defined as one logical block.

The file structure area 70 contains the micro UDF
And a management area defined by ISO9660, and through the description of this area, the video manager jar 71
Are stored in the system ROM / RAM unit 52. The video manager jar 71 describes information for managing a video title set, and has a file # 0.
Starting with a plurality of files 74. The video title set 72 includes compressed video data,
Sub-video data, audio data, and reproduction control information for reproducing these are recorded. Each video title set 72 is composed of a plurality of files 74. These files are also separated by logical sector boundaries.

In the other recording area 73, information used when using the information of the video title set or information independently used is recorded.

The video manager jar 71 will be described with reference to FIG.

The video manager jar 71 includes a video manager information (VMGI) 75 and a video object set (V
MGM VOBS) 76 and backup of video manager information (VMGI) (BUP) 77.

VMGM The VOBS 76 stores video data, audio data, and sub-picture data for a menu relating to the volume of the optical disc managed by the video manager 71. Audio and sub-picture descriptive information about each title in the volume,
A title selection display can be obtained. For example, if the optical disc contains English conversation for language learning, the title of the English conversation and the lesson example will be played back and displayed, the theme song will be played back as audio, and the level of the teaching material in the sub-video will be displayed. Is displayed. As a selection item, selection of a lesson number (level) is displayed, and the operation of the viewer is awaited. VMGM for such use VOBS 76 is used.

FIG. 15 shows a video object set (V
OBS) 82 is shown.

There are two types of video object sets (VOBS) for menus and one type for video titles, but both have the same structure.

Video Object Set (VOBS) 8
2 is defined as a set of one or more video objects (VOBs) 83, and the VOBs are used for the same purpose. Usually, a video object set (V
OBS) is configured as a video object (VOB) for displaying a plurality of menu screens, and a video object set (VOBS) for a video title set.
Is configured as a video object (VOB) for displaying a normal moving image or the like.

The video object (VOB) has an identification number (VOB). IDN # j), and the identification number (VOB A video object (VOB) can be specified using IDN # j). One video object (VOB) includes one or a plurality of cells 84. Similarly, the identification number (C ID
N # j), and the identification number (C IDN #
The cell can be specified using j). A video object for a menu may be composed of one cell.

Further, one cell is composed of one or a plurality of video object units (VOBU).
And one video object unit (VOBU)
Is defined as a pack sequence having one navigation pack (NV pack) at the top. One video object unit (VOBU) is stored in the NV pack (D
This is defined as a set of all packs recorded from the first NV (including SI) to immediately before the next NV pack.

Video Object Unit (VOBU)
Is equivalent to the playback time of video data composed of one or a plurality of GOPs (groups of pictures) included in the VOBU, and the playback time is about 0.4 seconds or more and within 1 second. Stipulated. According to the MPEG standard, one GOP compresses image data corresponding to a reproduction time of about 0.5 seconds. Therefore,
According to the MPEG standard, information for about 0.5 seconds is arranged for both audio and video.

One video object unit (VO
BU), the video pack (V pack), the sub-picture pack (SP pack),
Audio packs (A packs) are arranged. Therefore, a plurality of V packs in one VOBU have one GOP or a plurality of GO packs of compressed image data whose playback time is within one second.
The audio signal corresponding to the playback time is compressed and arranged as an A-pack. Also, the sub-picture data used during this playback time is compressed and arranged as an SP pack. However, the audio signal is recorded by packing data of, for example, eight streams and sub-pictures of, for example, 32 streams.

One stream of the audio signal is data encoded in one type of encoding format, and is composed of, for example, eight channels of linear PCM and 20-bit quantized data.

Returning to FIG. 14, the description will be continued.

Video Manager Information (VMGI) 75
Describes information for searching for a video title, and includes at least three tables 78, 79,
80 are included.

Volume management information management table (VMGI
MAT) is the size of the video manager (VMG) 71, the start address of each information in the video manager, and the video object set (VMGM) for the video manager menu. VOBS) is described.

Title search pointer table (TT)
SRPT) describes an entry program chain (EPGC) of a video title included in the volume of the optical disc, which can be selected according to the key operation of the apparatus and the input of the title number from the display unit 500.

The program chain will be described with reference to FIG. The program chain 87 is a set of program numbers for reproducing the story of a certain title, and the story chapter or story of one title is completed by continuously reproducing the program chain. One program number is composed of a plurality of cell identification numbers. The cell identification number is VO
A cell in the BS can be specified.

Video title set attribute table (VT)
S ART) 80 describes attribute information defined in a video title set (VTS) in the volume of the optical disc. The attribute information includes the number and number of VTSs, a video compression method, an audio encoding mode, a sub-video display type, and the like, and is described in the video title set attribute table.

As described above, in the case of the packet system according to the present invention, the audio data at the head of each packet is always the head of the sampling data, and can be handled in packet units. And its sequence processing becomes easy.

Next, an audio decoder for reproducing recorded data arranged as described above will be described.

FIG. 17 shows the basic configuration of the audio decoder 513.

This example shows an example of a decoder which can reproduce data in all modes of the number of channels and the number of bits of a sample shown in FIG. The input data shows a case where the number of quantization bits of all eight channels is 24 bits.

The input terminal 710 receives a series of samples described with reference to FIG. This sample sequence is provided to the input terminal 711 of the switch SW0. Switch SW
0 has a distribution terminal for each sample of channels An to Hn and an to hn. Terminals corresponding to the samples of each channel are given the same reference numerals as the representative samples. Here, samples A0 to H0, A1 to H1, a0 to h0, and a1 to h1 are shown as representative samples.

The terminals A0 to H0 and A1 to H1 have 16 bits, and the terminals a0 to h0 and a1 to h1 are 4-bit terminals. Since the extra sample may be 8 bits in total, two sets of 4-bit terminals a0 to h0 and a1 to h1 are prepared. The 16-bit terminal A0 is connected to the upper bit (16 bits) of the memory MA0, and the corresponding 4-bit terminal a
0 and a0 are stored in the memory M via switches J1 and J2, respectively.
It is connected to the lower bits (8 bits) of A0. 16
The bit terminal B0 is connected to the upper bit of the memory MB0 via the switch JB, and the corresponding 4-bit terminal b0, b
0 is connected to the lower bits of the memory MB0 via the corresponding switches j1 and j2. The 16-bit terminal C0 is connected to the upper bits of the memory MC0 via the switch JC, and the corresponding 4-bit terminals c0 and c0 are connected to the lower bits of the memory MC0 via the corresponding switches j1 and j2, respectively. Similarly, from each terminal D0 to H
0, D1 to H1, d0 to h0, and d1 to h1 are also connected to the corresponding memories MD0 to MH1, respectively.

Thus, each channel is stored in the memory MA0.
From MH1. Memory MA0
And the output terminal of MA1 are A channel output switch SWA
Terminals TA0, Ta0, Ta0, TA1, Ta1, Ta
1 connected. TA0 and TA1 are 16-bit terminals, and Ta0, Ta0, Ta1, and Ta1 are 4-bit terminals. Similarly, the output terminals of the memories MB0 and MB1 are connected to the terminal TB of the B-channel output switch SWB.
0, Tb0, Tb0, TB1, Tb1, Tb1. TB0 and TB1 are 16-bit terminals,
Tb0, Tb0, Tb1, and Tb1 are 4-bit terminals. Similarly, output terminals of other memories are connected to corresponding output switches.

Next, the operation will be described.

The samples S0, S1, e1, e2,... Arranged for recording / transmission input to the switch SW0 are A0, B0,.
, H1, a0, b0, ... h0, a1, b1
..., h0. Here, the main word of each channel is 16 bits, and the extra word is 8 bits. It is assumed that all open / close switches of the circuit are closed.
When the rotary switch SW0 is sequentially switched from the uppermost contact, the corresponding samples are transferred from the memories MA0 to MH1. As described above, by the operation of the rotary switch SW0, two pairs of samples are cyclically stored from the memories MA0 to MH1. Thereafter, of the samples stored in the memories MA0 to MH1, the sample of the desired channel is read out via the corresponding rotary switch. In the read sample, the main sample and the extra sample are decoded,
It is used after being synthesized.

Attention is paid to reading of channel A.
The rotary switch SWA first reads a 16-bit sample A0 at the uppermost 16-bit contact position. Next, a sample a0 of a total of 8 bits is read from two 4-bit contact positions. Further, a 16-bit sample A1 is read at the next 16-bit contact position. Next, a sample a1 of a total of 8 bits is read at two 4-bit contact positions. One rotation of rotating switch SWA, two pairs of channel A sample A
0, a0, A1, and a1 are read. Thus, two pairs of samples of channel A are obtained in time series. Thereafter, samples are read out for channels B, C,. Here, the rotary switch SW0,
Since SWA,... SWH each process two pairs of samples in one rotation, the rotation cycle is の of the sampling frequency.
(Fs / 2).

FIG. 18 shows another embodiment of the audio decoder.

This embodiment shows a state in which data is processed when the number of channels is 2 and the number of quantization bits of the sample is 20 bits. The difference from the circuit shown in FIG. 17 is the state of the switches JB-JH, j1, j2. Therefore, portions corresponding to the circuit in FIG. 17 are denoted by the same reference numerals.

When S0, S1, e0, e1,... Are represented by a sample sequence of each channel, A0, B0, A1, B1, a
0, b0, a1, b1,... Here, the main sample of each channel is 16 bits, and the extra sample is 4 bits.

As shown in the figure, the state of the switches JB... JH is such that only the switch JB is closed and JC.
It is open. As for the switches j1 and j2,
As shown, extra samples a0, b0, a1,
With the switch corresponding to b1, only j1 is closed and the others are open. In addition, other extra samples c0,
.. H0, c1,... H1 corresponding to switches j1, j2
Are all open.

In the portion where the switch is open, no data transfer is performed.

When the rotary switch SW0 separates the input data in synchronization with the data input, the data to be transferred becomes A
0, B0, A1, B1, a0 (4 bits), b0 (4 bits), a1 (4 bits), and b1 (4 bits).
By the operation of the rotation switch SW0, the memories MA0, MA0,
Samples are input only to MB0, MA1, and MB1 in the order shown in FIG.

On the output side, an output is obtained only from the memories of channels A and B among the memories MA0 to MH1. Data 0 is output from the memory of another channel. Of the switches j1 and j2 on the read side, j1 is closed and j2 is open. Therefore, a 4-bit extra sample is read after the 16-bit main sample. In the case of the channel A, when the switch SWA is switched, A0, a0
(4 bits), A1, and a1 (4 bits) in this order, and the data of channel A is sequentially output.

The setting and switching operation of each switch in the above-described embodiment are set programmable according to the number of channels of the audio stream and the number of quantization bits of the sample. Such a signal processing mode is described in the video title set attribute table shown in FIG. 14 and the pack header shown in FIG.
That is, the audio data included in the audio packet is linear PCM, and further, the audio frame number, the number of quantization bits, the sampling frequency,
An audio channel number and the like are described.

The decoder shown in FIG. 17 and FIG.
This is a so-called full decoder, which is applicable to high-end models capable of supporting all modes and reproducing all bits and all channels.

The concept of the present invention relates to a data arrangement, a recording / reproducing method and an apparatus which can support various combinations of the number of channels and the number of quantization bits. Simple models that require low cost, as well as being able to support high-end models such as those described above,
This data array is compatible with models that reproduce only 2 channels and 16 bits for all modes. Such a model requires a smaller circuit scale than a high-end model.

In the above figures and description, the switches used for sorting each sample and taking out the samples from the memory are mechanically shown. However, these are all constituted by electronic circuit means.

Next, an audio decoder in a simple reproduction model will be described. This audio decoder
The decoder processes 16-bit data of only the channels A and B. It is assumed that the input samples are 8 channels and the quantization bits are 24 bits.

In FIG. 19, the sequence of samples described with reference to FIG. 1 is continuously input to the input terminal 810. This sample sequence is given to the input terminal 811 of the switch SW0. The switch SW0 is connected to channels An to Hn, an
To hn for each sample. Terminals corresponding to the samples of each channel are given the same reference numerals as the representative samples. Here, as a representative sample,
Samples A0 to H0, A1 to H1, a0 to h0,
a1 to h1 are shown.

The terminals A0 to H0 and A1 to H1 have 16 bits, and the terminals a0 to h0 and a1 to h1 are 4-bit terminals. Since the extra sample may be 8 bits in total, two sets of 4-bit terminals a0 to h0 and a1 to h1 are prepared.

However, in this decoder, terminals A0 and B
Only 0, A1, and B1 are connected to the memories MA and MB, and the other terminals C0-H0 and c0-h0 are grounded. The switch SW0 may be manufactured in this way, or only the system related to the channels A and B may be manufactured from the beginning.

The switches SWA and SWB for reading data from the memories MA and MB are switches for reading data in 16-bit units. These switches SWA, SWB
Are operated to obtain output data matching.

The operation will be described.

The samples S0, S1, e0, e1,... Of the recording or transmission array input to the switch SW0.
Are represented by A0, B0, ...,
H0, A1, B1,..., H1, a0, b0,.
.., H0. Here, the main sample of each channel is 16 bits, and the extra sample is 8 bits. All open / close switches of the circuit are closed. When the rotary switch SW0 is sequentially switched from the uppermost contact point, the corresponding samples are transferred from the memory MA to the MB1. All other samples are rejected.

Thereafter, the samples stored in the memories MA0 and MB1 are read as the samples of the channels A and B.

Since the rotation switch SW0 processes two samples in one rotation, it is の of the sampling frequency fs. The rotation switches SWA and SWB are set to 1 for one rotation.
Since the sample is read, the frequency is fs.

Next, another audio decoder in a simple reproduction model will be described. This audio decoder is a decoder that processes 16-bit data of only channels A and B. It is assumed that the input samples are two channels and the quantization bits are 20 bits.

In FIG. 20, the sequence of the samples described with reference to FIG. This sample sequence is given to the input terminal 811 of the switch SW0. The switch SW0 is connected to channels An to Hn, an
To hn for each sample. Terminals corresponding to the samples of each channel are given the same reference numerals as the representative samples. Here, as a representative sample,
Samples A0, B0, A1, B1, a0, b0, a1,
b1 is shown.

The terminals A0, B0, A1, and B1 have 16 bits, and the terminals a0, b0, a1, and b1 are 4-bit terminals. 2 channels, 20 quantization bits
To accommodate the bit mode, only switch JB is closed and switches JC-JH are open. Switches j1, j2 corresponding to terminals a0, b0, a1, b1
Are closed, and switches j3-j16 corresponding to the other terminals are open.

When the rotary switch SW0 is sequentially switched in the above state, data transfer is not performed in the open switch. And the main samples A0, B0, A
Only 1 and B1 are children to be transferred to the memories MA and MB. Also, extra samples aa0, b0, a1, b1
Is discarded because the corresponding switch is grounded. The operation of reading samples from the memories MA and MB is performed in the same manner as in the previous embodiment.

In the example of the simple model described above, the case of two types of modes has been described. However, data of two channels in all modes can be taken out according to the open / closed state of the switch. Of particular note is that the processing for extra samples is in 8-bit units. With such a data arrangement, the number of bits of a pair of extra samples becomes an integral multiple of 8 bits regardless of the number of channels, even if the extra word of each channel is 4 bits. For this reason, even when the extra samples are rejected in the simple decoder, the processing can be performed in units of 8 bits.

Since the main word of the main sample is 16 bits, all the processing can be performed in 8-bit units, and there are many advantages in forming a specific circuit.

FIG. 21 schematically shows a pack header of an audio pack.

First, the pack start code (4 bytes)
And then the system clock reference (SCR)
Is described. System clock reference (S
(CR) indicates the loading time of this pack. If the value of this SCR is smaller than the value of the reference time inside the apparatus, the pack to which this SCR is added is loaded into the audio buffer. In the pack header, the program multiplexing rate is described in 3 bytes. Further, the stuffing length is also described in one byte.
The control circuit can determine the read address of the control information by referring to the stuffing length by the control circuit.

FIG. 22 shows the contents of the packet header of the audio packet. The packet header includes a packet start code prefix for notifying the start of the packet, a stream ID indicating what data the packet has, and data indicating the length of the packet stream. Various types of information of the packet elementary stream (PES), such as a flag indicating prohibition and permission of copying, a flag indicating original information or copied information, and a length of a packet header are described. Furthermore, a presentation time stamp (PTS) for synchronizing the output of the packet with another video or sub-picture is described. Further, in the first packet of the first field in each video object, information such as a flag indicating whether or not a buffer is described and the size of the buffer are described.

Further, it has a stuffing byte of 0-7 bytes.

Further, it has an audio stream, a linear PCM or another compression system, and a substream ID for indicating the number of the audio stream.
Furthermore, the number of audio frames in which the first byte is arranged in this packet is described. Furthermore, a pointer indicating the first audio frame in the packet to be reproduced at the time pointed by the PTS, that is, the first byte of the unit to be accessed first is described. This pointer is described by the byte number from the last byte of this information. The pointer indicates the first byte address of the audio frame. In addition, an audio enhancement flag indicating whether or not high-frequency enhancement has been performed, a mute flag for obtaining mute when audio frame data is all 0, and a frame number to be accessed first in an audio frame group (GOF) are also described. Have been.
Also, the length of the quantization word, that is, the number of quantization bits, the sampling frequency, the number of channels, control information of the dynamic range, and the like are described.

The above header information is analyzed by a decoder control unit (not shown) in the audio decoder.
The decoder control unit switches the signal processing circuit of the decoder to a signal processing mode corresponding to the audio data currently being captured. FIGS. 17 to 20 show the switched state.
It is as shown by. Since the same information as the above-mentioned header information is also described in the video manager, if such information is read at the beginning of the reproducing operation, it is not necessary to read the same sub-stream thereafter. However, as described above, the mode information necessary to reproduce audio is described in the header of each packet because, for example, when a packet sequence is transmitted in a communication sequence, reception is started at any time. This is because the receiving terminal can recognize the audio mode. Also, this is because audio information can be reproduced even when only the pack is taken in by the audio decoder.

FIG. 23 shows a signal sequence of a reproducing apparatus relating to an audio stream. 23, the inside of the system processing unit 504 and the inside of the audio decoder 513 are shown in detail in the apparatus of FIG.

The high-frequency signal (read signal) input to the system processing unit 504 is input to the synchronization detector 601. The synchronization detector 601 detects a synchronization signal added to the recording data and generates a timing signal. The read signal from which the synchronization signal has been removed by the synchronization detector 601 is
8-16 demodulator 602 for demodulating 16 bits to 8 bits
And demodulated into an 8-bit data string. The demodulated data is input to the error correction circuit 603, where the data is subjected to error correction processing. The error-corrected data is input to the demultiplexer 604. In the demultiplexer 604, an audio pack, a video pack, and a sub-picture pack are identified based on the stream ID, and each pack is output to a corresponding decoder.

The audio pack is taken into the audio buffer 611. The pack header and packet header of the audio pack are stored in the control circuit 612.
Is read. The control circuit 612 recognizes the contents of the audio pack. That is, the start code, stuffing length, packet start code, and stream ID of the audio pack are recognized. Recognition of packet length, substream ID, recognition of initial access point, recognition of audio quantization bit number,
It also recognizes the number of channels and the sampling frequency. When such information is recognized, the stuffing byte length and the padding packet length are determined from the table shown in FIG. The control circuit 612
Recognizes a linear PCM packet based on the substream ID.

As a result, the control circuit 612 can grasp the cut-out address of the audio data stored in the audio buffer 611.
Therefore, the audio buffer 611 is controlled by the control circuit 612, and the sample described above,
For example, S0, S1, e0, e1, S2, S3,... Can be output. The control circuit 612 can recognize the number of stuffing bytes and the number of padding packets by recognizing at least the number of quantization bits, the sampling frequency, and the number of audio channels. Then, based on the recognition information, the data can be cut out.

This sample is supplied to the channel processor 613.
Supplied to The inside of this channel processor 613
The operation mode is controlled by the control circuit 612 as described with reference to FIGS.

Next, the physical relationship between the audio packets, video packets, and sub-picture packets described above and the recording tracks of the optical disk will be described.

FIGS. 24 (A), 24 (B), 24
As shown in (C) and FIG. 24 (D), when the recording surface of a part of the optical disc 10 is enlarged, a pit row is formed. This set of pits forms a sector. Therefore, a sector row is formed on the track of the optical disc. This sector is read continuously by the optical head. Then, the audio pack is reproduced in real time.

Next, one sector, for example, a sector in which audio information is described will be described. As shown in FIGS. 25A and 25B, one sector is composed of 13 × 2 frames. Then, a synchronization code is added to each frame. In the drawing, the arrangement of the frames is shown two-dimensionally, but they are recorded on the track in order from the first frame. SY0, SY5, SY1, SY5, SY5, SY5, SY5
SY2, SY5,...

The number of bits of the synchronization code and data in one frame shown in the figure is 32 bits and 1456 bits. 32 bits = 16 bits × 2, 1456 bits = 16 bits × 91. This equation means that a 16-bit modulation code is recorded. This is because, when recording is performed on the optical disc, 8-bit data is modulated into 16 bits and recorded.
Further, the sector information includes a modulated error correction code.

FIG. 26A shows one of the above physical sectors.
This shows one recording sector after 6-bit data is decoded into 8 bits. The data amount of this recording sector is
(172 + 10) bytes × (12 + 1) lines.
Each line is provided with a 10-byte error correction code. Also, there is an error correction code for one line, and this error correction code functions as an error correction code in the column direction when 12 lines are collected, as described later.

When the error correction code is removed from the data of one recording sector, a data block as shown in FIG. 26B is obtained. That is, a 6-byte sector ID, a 2-byte ID error detection code, and a 6-byte copyright management information are added to the beginning of the data to the 2048-byte main data, and a 4-byte error detection code is added to the end of the data. Data block.

The above 2048-byte data is one pack described above, and a pack header, a packet header, and audio data are described from the beginning of this one pack. Then, various guide information for processing the audio data is described in the pack header and the packet header.

As described above, one packet in which audio samples are arranged is allocated to one sector of the disk and recorded. Then, even if the information is information of one sector, the audio decoder
CM data can be favorably reproduced. This is 1
This is because the beginning of the audio data in the pack is always distributed so as to start from the beginning of the main sample. Also, this is because the pack header and the packet header describe control information sufficient for the audio decoder to process the audio data.

Next, an error correction code block (ECC block) will be described.

As shown in FIGS. 27 (A) and 27 (B), the ECC block is constituted by a set of 16 recording sectors described above. FIG. 27 (A)
Data sector of 12 rows x 127 bytes (Fig. 26 (A))
Shows a state in which 16 are collected. Then, a 16-byte outer code parity (PO) is added to each column. Each row has a 10-byte inner code parity (PI)
Is added. Furthermore, before recording, FIG.
As shown in (B), the outer code parity (PO) of 16 bytes is distributed to each row by 1 bit. As a result,
One recording sector is configured as data of 13 (= 12 + 1) rows. In FIG. 27A, B0,
.., 0, B0, 1,... Indicate addresses in byte units. In FIG. 27 (B), 0 to 15 assigned to each block represent one recording sector.

On the recording track of the above-mentioned disc,
Video pack, sub-picture pack, audio pack, NV
The packs are interleaved and arranged. However, the invention is not limited to such disks. The present invention can also be applied to a disc in which only the audio pack rows are recorded. Also, the present invention can be applied to a disc in which an audio pack and a sub-picture pack are interleaved. Also, the present invention can be applied to a disc in which an audio pack, a sub-picture pack, and an NV pack are interleaved. These combinations are free.

[0150]

As described above, according to the present invention,
Other channel signals of higher quality than conventional CDs can be recorded, and reproduction processing is easy for both simple models and higher-order models, which is effective for recording or processing linear PCM data or the like.

[Brief description of the drawings]

FIG. 1 is an explanatory view showing a sample configuration and a sample arrangement shown for explaining a basic embodiment of the present invention.

FIG. 2 is an explanatory diagram showing a relationship between a sample, a frame, and a group in FIG. 1;

FIG. 3 is an explanatory diagram showing a relationship between an audio frame and a pack sequence according to the present invention.

FIG. 4 is a diagram showing a generalized audio data array in a 20-bit mode and a 24-bit mode.

FIG. 5 is an explanatory diagram showing the principle of interleaving.

FIG. 6 is an explanatory diagram showing an example of an arrangement of packs according to the present invention and a configuration of an audio pack in the arrangement.

FIG. 7 is a diagram illustrating the configuration of an audio pack in detail.

FIG. 8 is an explanatory diagram showing an example of a data size in a packet of linear PCM data to which the present invention is applied;

FIG. 9 is an explanatory diagram showing a procedure for generating an audio pack.

FIG. 10 is a block diagram of a disk reproducing apparatus.

FIG. 11 is an explanatory diagram of a disk drive unit.

FIG. 12 is an explanatory diagram of an optical disk.

FIG. 13 is an explanatory diagram showing a logical format of an optical disc.

FIG. 14 is an explanatory diagram of the video manager of FIG. 13;

FIG. 15 is an explanatory diagram of the video object set in FIG. 14;

FIG. 16 is an explanatory diagram of a program chain.

FIG. 17 is a diagram showing an example of a basic circuit configuration of an audio decoder according to the present invention.

FIG. 18 is a diagram showing another example of the basic circuit configuration of the audio decoder according to the present invention.

FIG. 19 is a diagram showing another example of the basic circuit configuration of the audio decoder according to the present invention.

FIG. 20 is a diagram showing still another example of the basic circuit configuration of the audio decoder according to the present invention.

FIG. 21 is a diagram showing contents of a pack header of an audio pack.

FIG. 22 is a diagram showing contents of a packet header of an audio pack.

FIG. 23 is a block diagram showing the configuration of an audio processing system of the disc reproducing apparatus.

FIG. 24 is an illustration showing a disk, a pit row, a sector row, and a physical sector.

FIG. 25 shows the contents of a physical sector.

FIG. 26 is a diagram showing a configuration of a recording sector.

FIG. 27 is a diagram showing a configuration of an error correction code block.

[Explanation of symbols]

 Reference Signs List 10 disk, 501 disk drive unit, 502 system CPU, 503 system ROM / RAM, 504 system processing unit, 505 data RAM, 506 video buffer, 507 sub-picture buffer, 508 video decoder, 509 .., A sub-picture decoder, 511, a D / A converter, 512, an audio buffer, 513, an audio decoder, 514, a D / A converter.

Continuation of the front page (72) Inventor Hirohisa Nishiwaki 70, Yanagicho, Saiwai-ku, Kawasaki-shi, Kanagawa Prefecture Inside the Toshiba Yanagicho Plant (56) References JP-A-9-251717 (JP, A) JP-A-9-252449 ( JP, A) JP-A-9-251723 (JP, A) JP-A-56-21446 (JP, A) JP-A-57-207960 (JP, A) JP-A-60-201571 (JP, A) JP-A-5-20794 (JP, A) JP-A-6-162671 (JP, A) JP-A-8-31104 (JP, A) JP-A-55-52514 (JP, A) JP-A-1-282779 (JP) JP-A-4-176060 (JP, A) JP-A-5-250811 (JP, A) JP-A-8-63900 (JP, A) JP-A-8-63901 (JP, A) 9-120645 (JP, A) JP-A-6-76473 (JP, A) JP-A-8-22674 (JP, A) JP-A-4-101956 (JP, U) (58) Fields investigated (Int. Cl. ⁶ , DB name) G11B 20/10-20/12 103

Claims (21)

(57) [Claims] 1. A signal of one or more channels
Samples in time series and processes the quantized data
In this case, a sample consisting of M bits of the signal of each channel
The pull data is connected to the m1 bit main word on the MSB side.
Divided into m2 bit extra words on the LSB side,
Main of (2n) th sample data of each channel
Words are put together and placed as main sample S2n.
Followed by the (2n + 1) th sample data of each channel
The main word of the data into a main sample S2n + 1
And then the (2n) th sample of each channel
Extra word for pull data
Sample e2n, and then (2) of each channel.
The extra word of the (n + 1) th sample data is stored.
And placed as extra sample e2n + 1 (where n
= 0, 1, 2, ...)And Copy the sample arranged as above to the pack header and packet
Packet header, limited to specified bytes with data part
Packed in the data part in the pack, the pack header contains
Structure with information including the presence and length of stuffing added
And the data of this structure Record or transmit on a recording medium
Data for recording or transmission characterized by the fact that
Processing method. 2. A system for sampling one channel or a plurality of channels in time series, and recording or transmitting quantized data to reproduce the sampled data comprising M bits of signal data of each channel. , The main word of m1 bit on the MSB side and the extra word of m2 bit on the LSB side, and the main word of the (2n) th sample data of each channel is collectively arranged as a main sample S2n. The main word of the (2n + 1−2k) th sample data of the channel is put together and the main sample S
2n + 1-2k, and then extra words of (2n) -th sample data of each channel are put together to form an extra sample e2n
Then, the extra words of the (2n + 1−2k) th sample data of each channel are put together and arranged as an extra sample e2n + 1−2k (where n = 0, 1, 2, 2).
.. And k = constant integer), and the samples arranged as described above are packed header and packet.
Packet header, limited to specified bytes with data part
Packed in the data part in the packet,
Has information including the presence and length of stuffing.
A method for processing data for data recording or transmission, characterized by recording or transmitting data having this structure on a recording medium. 3. The m1 is 16 bits, and m2 is 4 bits.
3. The data processing method according to claim 1, wherein the data is a bit. 4. The m1 is 16 bits, and m2 is 8 bits.
3. The data processing method according to claim 1, wherein the data is a bit. 5. The method according to claim 1, wherein the placement data is the main sample.
A unit of S2n, S2n + 1 and extra words e2n, e2n + 1 is a unit. One frame is formed by collecting a predetermined number of units, and one group is formed by collecting a predetermined number of frames. 2. The data processing method according to claim 1, wherein: 6. The method according to claim 6, wherein the placement data is the main sample.
A unit of S2n, S2n + 1 and extra words e2n, e2n + 1 is defined as a unit. One frame is formed by collecting a predetermined number of units, and each frame is composed of a plurality of packs.
2. The data processing method according to claim 1, wherein the plurality of packs are allocated to a data unit , and the plurality of packs are mixed with video packets and sub-picture packets, and are arranged between control packets. Wherein said arrangement data further the number of samples of a given period are one frame, the frame sequence, the data unit of one pack in the pack of limited pack sequence a predetermined number of bytes To the packet header
2. A data processing method according to claim 1, wherein said data has a structure including at least quantization bit information and channel information of said quantized data. Wherein said pack is a predetermined byte length, the
When arranging a plurality of main and extra samples in the data section , other samples are sequentially arranged at the predetermined position of the data section by aligning the beginning of the first main sample, and the total of the plurality of main and extra samples is calculated. The byte length should be equal to or less than the maximum allowable byte length of the data portion. If the total byte length is less than the maximum byte length, stuffing bytes or padding is performed to compensate for the extra portion. 7. The data processing method according to claim 6, wherein invalid data by a packet is inserted. 9. The data processing method according to claim 8, wherein the plurality of main and extra samples are linear PCM data, and the maximum byte length is 2010 bytes. In 10. If the total byte length is less than said maximum byte length, if this excess length is less than or equal to 7 bytes are inserted the stuffing bytes into the packet header, in the case of more than 8 bytes 9. The data processing method according to claim 8, wherein a padding packet is inserted at a rear part of the data part . 11. When one-channel or plural-channel signals are sampled in time series and quantized data is processed, sample data consisting of M bits of each channel signal is converted to an m1 bit main word of the MSB side. And the m2 bit extra word on the LSB side, the main word of the (2n) th sample data of each channel is collectively arranged as a main sample S2n, and then the (2n + 1) th sample data of each channel The main word of the main sample S
2n + 1, then extra words of (2n) -th sample data of each channel are collectively arranged as extra samples e2n, and then extra words of (2n + 1) -th sample data of each channel are It is arrangement data of a structure arranged as an extra sample e2n + 1 (where n = 0, 1, 2,...), And the arrangement data is the main sample S2n, S2n + 1, extra sample e2n, e2n + and 1 chunk is a unit, one frame by a predetermined number of units gather is formed, chronologically consecutive frames column, Pakkuhe'
, A packet sequence having a packet header and a data portion.
The pack is allocated to a data portion , the pack is a predetermined byte length, and when arranging a plurality of main and extra samples in the data portion , the head of the first main sample is aligned with a predetermined position of the data portion. samples are sequentially arrayed in a total byte length of said main and extra samples of said plurality of front
To less than the maximum byte length of serial data unit is acceptable, when said total byte length is less than said maximum byte length, the insertion of invalid data caused by stuffing bytes or padding packets to compensate for the excess length How to process the featured data. 12. When one-channel or plural-channel signals are sampled in time series and quantized data is processed, M-bit sample data of each channel signal is converted to an m1 bit main word on the MSB side. And the m2 bit extra word on the LSB side, the main word of the (2n) th sample data of each channel is collectively arranged as a main sample S2n, and then the (2n + 1) th sample data of each channel The main word is put together and the main sample S2n
+1. Then, the extra words of the (2n) th sample data of each channel are put together and arranged as extra samples e2n. Then, the extra words of the (2n + 1) th sample data of each channel are put together. arranged as an extra sample e2n + 1 Te (however, n = 0,1,2, ...), and it is arranged so as of this
Sample packed header, packet header, data part
Said data portion in a pack restricted to a predetermined byte having
And the pack header has stuffing.
A structure to which information including the length is added is used.
Device that plays back stuffing data by referring to stuffing information.
A signal processing unit for removing a data and extracting only at least the m1 bit main word from the arrangement data to obtain a reproduction output. 13. The m channel on the MSB side of a predetermined channel.
1-bit main word and LS of the corresponding channel
13. The signal processing apparatus according to claim 12, further comprising means for obtaining a reproduction output by combining the m-bit extra word on the B side. 14. The signal processing apparatus according to claim 13, wherein said m1 processed by said means for obtaining a reproduction output is 16 bits, and m2 is 4 bits. 15. The signal processing apparatus according to claim 13, wherein said m1 processed by said means for obtaining a reproduction output is 16 bits, and m2 is 8 bits. 16. When one channel or a plurality of channels of signals are sampled in time series and quantized data is processed, sample data consisting of M bits of each channel signal is converted to a main word of m1 bits on the MSB side. And the m2 bit extra word on the LSB side, the main word of the (2n) th sample data of each channel is collectively arranged as a main sample S2n, and then the (2n + 1) th sample data of each channel The main word of the main sample S
2n + 1, then extra words of (2n) -th sample data of each channel are collectively arranged as extra samples e2n, and then extra words of (2n + 1) -th sample data of each channel are It is arrangement data of a structure arranged as an extra sample e2n + 1 (where n = 0, 1, 2,...), Wherein the arrangement data is the main sample S2n, S2n + 1, extra word e2n, e2n + and 1 chunk is a unit, one frame by a predetermined number of units gather is formed, chronologically consecutive frames column, Pakkuhe'
, A packet sequence having a packet header and a data portion.
An apparatus for reproducing the pack, which is allocated to the data section , and refers to the stuffing information to perform the stuffing data.
Input means for removing words from each of the channels while removing the data, a plurality of memories for storing the words from the input means, and a plurality of output means for deriving the words of each channel from the memories of the respective channels. A signal processing device comprising: 17. The input means for allocating transmitted words of at least two channels, the plurality of memories storing the words of the two channels, and the output means storing the words of the two channels. 2. The method of claim 1, further comprising deriving a word of the stored memory.
7. The signal processing device according to 6. 18. The packet header includes at least quantization bit information and channel information of the quantized data, and recognizes and reproduces the quantization bit information and channel information included in the packet header. 2. The apparatus according to claim 1, further comprising means for setting a mode.
7. The signal processing device according to 6. 19. A signal obtained by sampling a signal of one channel or a plurality of channels in time series and quantizing it.
In a system for recording or transmitting and reproducing data, sample data consisting of M bits of a signal of each channel is divided into an m1 bit main word on the MSB side and an m2 bit extra word on the LSB side. The main words of the (2n) -th sample data are put together and arranged as a main sample S2n. Then, the main words of the (2n + 1-2k) -th sample data of each channel are put together and put into the main sample S2n.
2n + 1-2k, and then extra words of (2n) -th sample data of each channel are put together to form an extra sample e2n
Then, the extra words of the (2n + 1−2k) th sample data of each channel are put together and arranged as an extra sample e2n + 1−2k (where n = 0, 1, 2, 2).
... and k = a constant integer), is arranged so that this
Packed sample, packed header, packet header, data
Said data in a pack limited to a given byte with
Stuffing on the pack header.
A structure to which information including nothing and length is added, and
Device that plays back the stuffing data by referring to the stuffing information.
A signal processing device for removing only the m1 bit main word of the predetermined channel by removing data. 20. When one channel or a plurality of channels of signals are sampled in time series and quantized data is processed, sample data consisting of M bits of each channel signal is converted to an m1 bit main word of the MSB side. And the m2 bit extra word on the LSB side, the main word of the (2n) th sample data of each channel is collectively arranged as a main sample S2n, and then the (2n + 1-2k) th Combine the main words of the sample data into the main sample S
2n + 1-2k, and then extra words of (2n) -th sample data of each channel are put together to form an extra sample e2n
Then, the extra words of the (2n + 1−2k) th sample data of each channel are put together and arranged as an extra sample e2n + 1−2k (where n = 0, 1, 2, 2).
... and k = a constant integer), is arranged so that this
Packed sample, packed header, packet header, data
Said data in a pack limited to a given byte with
Stuffing on the pack header.
A structure to which information including nothing and length is added, and
Device that plays back the stuffing data by referring to the stuffing information.
Data, and the m of a predetermined channel of the data section is removed.
A signal processing apparatus comprising means for combining a 1-bit main word with the m2-bit extra word of a corresponding channel to obtain a reproduction output. 21. A signal of one channel or a plurality of channels.
Signals in time series and process the quantized data.
In the case of processing, the
The sample data into the MSB side m1 bit main word
And m2 bit extra words on the LSB side
Of the (2n) th sample data of each channel
Collect inwords and place them as main sample S2n
Then, the (2n + 1) th sump of each channel
The main word of the data is put together and the main sample S2n
+1 and then (2n) -th of each channel
Extract extra words of sample data of
Placed as a tiger sample e2n, followed by each channel
Of the (2n + 1) th sample data of
And put them together as an extra sample e2n + 1 (however,
, N = 0, 1, 2,...)
The number of samples for a given period of the data structure of
And this frame sequence isPack header, packet header, data
Pack of packs including data part and limited to a specified number of bytes
Assigned to the data in the column, In the packet header Quantizing the quantized data
At least bit information and channel information are included.
The pack header contains the presence and length of stuffing.
Configuration data with additional informationReceiving terminal to receive
DeviceStuffing information contained in the pack header
And remove stuffing data, Previous
Bit information and channel information included in the header
Means for recognizing file information and setting the playback mode.
And a receiving terminal device.