JP2000214894A - Sound coding device, record medium, sound decoding device, sound transmitting method and transmission medium - Google Patents

Abstract

PROBLEM TO BE SOLVED: To improve decoding efficiency on the reproducing side, when multichannel sound signals are coded at a variable compression rate. SOLUTION: An audio data area in an audio packet is constituted of a plurality of PPCM access units, and each PPCM access unit is constituted of PPCM sink information and a sub-packet. The PPCM sink information includes the number of samples per packet (40, 80, or 160, depending on the sampling frequency), a data rate ['0', when it is VBR(variable bit rate): an identifier showing that data in the sub-packet are compressed data], sampling frequency and the number of quantizing bits, channel allocating information, and the like.

Description

DETAILED DESCRIPTION OF THE INVENTION

[0001]

BACKGROUND OF THE INVENTION 1. Field of the Invention The present invention relates to an audio encoding apparatus for compressing a multi-channel audio signal, a recording medium,
The present invention relates to an audio decoding device, an audio transmission method, and a transmission medium.

[0002]

2. Description of the Related Art As a method for compressing an audio signal, the present inventor has disclosed in Japanese Patent Application No.
A plurality of predictors having different characteristics are used to calculate a plurality of linear prediction values of a current signal from a past signal in a time domain for an original digital audio signal of a channel, and the original digital audio signal and the plurality of linear prediction values are calculated. A prediction coding method for calculating a prediction residual for each predictor from, and selecting a minimum value of the prediction residual is proposed.

In the above method, the original digital audio signal has a sampling frequency = 96 kHz and the number of quantization bits = 2.
Although a certain degree of compression effect can be obtained in the case of about 0 bits, in recent DVD audio discs, this 2
A double sampling frequency (= 192 kHz) is used, and the number of quantization bits tends to be 24 bits. Further, the sampling frequency and the number of quantization bits in the multi-channel may be different for each channel.

[0004]

By the way, since a compression rate such as a predictive coding method has a variable compression ratio (VBR: variable bit rate), when predictive coding of a multi-channel audio signal is performed, Of data greatly changes over time. When transmitting such data, the data is transmitted as a data stream instead of parallel for each channel. Therefore, it is necessary for the reproduction side (decoding side) to be able to reproduce (presentation) such a variable-length data stream in synchronization with each channel.

[0005] Therefore, the present invention provides an audio encoding device, a recording medium, an audio decoding device, and an audio transmission method that can improve decoding efficiency on the reproduction side when encoding a multi-channel audio signal at a variable compression rate. And a transmission medium.

[0006]

The present invention comprises the following means (1) to (9) to achieve the above object. That is,

1) Compressing means for compressing a multi-channel audio signal having a certain sampling frequency and the number of quantization bits for each channel by a predictive coding method, and a subpacket including data for each channel compressed by the compressing means. Means for formatting into a data structure having a synchronization information section including the sampling frequency and the number of quantization bits. 2) An audio coding apparatus according to claim 1, wherein the data structure is formatted by adding a header including SCR information corresponding to a sub packet and a synchronization information section. 3) A multi-channel audio signal having a certain sampling frequency and a certain number of quantization bits is compressed by a predictive coding method for each channel, a subpacket including the compressed data for each channel, and the sampling frequency and the number of quantization bits. A recording medium that has been formatted and recorded in a data structure having a synchronization information section including: (4) The recording medium according to (3), wherein the data structure is formatted by adding a header including SCR information corresponding to a sub-packet and a synchronization information part. 5) A multi-channel audio signal having a certain sampling frequency and a certain number of quantization bits is compressed by a predictive coding method for each channel, and a subpacket including the compressed data for each channel, and the sampling frequency and the number of quantization bits A speech decoding apparatus for decoding a data structure having a synchronization information section including: a means for separating the data structure into subpackets and a synchronization information section; and a decompression means for expanding compressed data in the subpacket for each channel. And a means for converting the expanded audio data into an analog audio signal based on the sampling frequency and the number of quantization bits in the synchronization information section. 6) A multi-channel audio signal having a certain sampling frequency and a number of quantization bits is compressed for each channel by a predictive coding method, a subpacket including the compressed data for each channel, and the sampling frequency and the number of quantization bits. And a header including a sub-packet and a header including SCR information corresponding to the sub-packet and the synchronizing information section, wherein the first decoder separates the SCR information included in the header. Separating means, a buffer for holding the sub-packet and synchronization information part based on the separated SCR information, and second separating means for separating the sub-packet and synchronization information part held in the buffer Decompression means for decompressing the compressed data in the subpacket for each channel based on the identifier in the synchronization information section, And means for converting the analog audio signal based on the serial decompressed audio data to a sampling frequency and number of quantization bits in the synchronization information unit includes speech decoding apparatus. 7) A multi-channel audio signal having a certain sampling frequency and a certain number of quantization bits is compressed by a predictive coding method for each channel, a subpacket including the compressed data for each channel, and the sampling frequency and the number of quantization bits. And transmitting a packet having a data structure formatted into a data structure having a synchronization information section through a communication line. (8) The voice transmission method according to (7), wherein the data structure is formatted by adding a header including SCR information corresponding to a sub-packet and a synchronization information section. 9) A multi-channel audio signal having a certain sampling frequency and a certain number of quantization bits is compressed by a predictive coding method for each channel, a subpacket including the compressed data for each channel, and the sampling frequency and the number of quantization bits. A transmission medium for transmitting a packet having a data structure formatted into a data structure having a synchronization information section including:

[0008]

Embodiments of the present invention will be described below with reference to the drawings. FIG. 1 is a block diagram showing a first embodiment of a speech encoding apparatus and a speech decoding apparatus according to the present invention, FIG. 2 is a block diagram showing the encoding unit of FIG. 1 in detail, and FIG. FIG. 4 is an explanatory diagram showing the format of a DVD pack, FIG. 5 is an explanatory diagram showing the format of a DVD audio pack, and FIG. 6 is an explanatory diagram showing the format of a DVD audio pack. Explanatory diagram showing the format of the area in detail, FIG.
FIG. 8 is a block diagram showing the decoding unit of FIG. 1 in detail, and FIG.
9 is a timing chart showing write / read timing of the input buffer, FIG. 9 is an explanatory diagram showing the amount of compressed data for each access unit, and FIG. 10 is an explanatory diagram showing an access unit and a presentation unit.

Here, as the multi-channel system, for example, the following four systems are known. (1) Four-channel system Like the Dolby surround system, a total of four channels of three channels of front L, C, and R + one channel of rear S. (2) Five-channel system S of Dolby AC-3 system
As without W channel, 3 channels of front L, C and R + 2 channels of rear SL and SR, total 5 channels (3) 6 channel system DTS (Digital Theater)
6) (L, C, R, SW (Lfe), SL, SR) such as the Dolby AC-3 system (4) 8-channel system SDDS (Sony Dynamic D
digital sound), forward L, LC, C, R
6 channels of C, R, SW + 2 channels of rear SL, SR, total 8 channels

[0010] The 6-channel (ch) mix & matrix circuit 1 'on the encoding side shown in FIG. 1 includes a front left (Lf), a center (C), a front right (Rf), a surround left ( Ls), surround light (Rs) and Lfe (Low Frequency Effe)
ct), the 6-channel PCM data is classified and converted into 2ch “1” and “2” for the front group and 4ch “3” to “6” for the other group by the following equation (1), and converted into 2ch “1”.
"2" is assigned to the first encoding unit 2'-1, and 4ch "3" to
"6" is output to the second encoding unit 2'-2. “1” = Lf + Rf “2” = Lf−Rf “3” = C− (Ls + Rs) / 2 “4” = Ls + Rs “5” = Ls−Rs “6” = Lfe−a × C where 0 ≦ a ≦ 1 ... (1)

As shown in detail in FIG. 2, the first and second encoders 2'-1 and 2'-2 which constitute the encoder 2 'respectively have 2ch "1", "2" and 4ch "3". "~" 6 "PC
The M data is predictively encoded for each channel, and the encoded prediction data is transmitted as a bit stream as shown in FIG. 3 to the decoding side via a recording medium 5 or a communication medium 6 such as a satellite line or a telephone line. On the decoding side, the first and second decoding units 3'-1 and 3'-2 constituting the decoding unit 3 'respectively perform 2ch "1" for the forward group, as shown in detail in FIG.
The prediction coded data of "2" and 4ch "3" to "6" relating to the other groups are decoded into PCM data for each channel.

Then, the original 6 ch (Lf, C, Rf, Ls,
Rs, Lfe) are restored, and stereo 2-ch data (L, R) is generated from the original 6 ch and coefficients mij (i = 1, 2, j = 1, 2 to 6) as in the following equation (2). I do. L = m11 · Lf + m12 · Rf + m13 · C + m14 · Ls + m15 · Rs + m16 · Lfe R = m21 · Lf + m22 · Rf + m23 · C + m24 · Ls + m25 · Rs + m26 · Lfe (2)

Referring to FIG. 2, encoding sections 2'-1, 2'-
2 will be described in detail. PC for each channel "1" to "6"
The M data is stored in one frame buffer 10 for each frame. Then, the sample data of each of the channels “1” to “6” of one frame are respectively supplied to the prediction circuits 13D1 and 13D.
2, 15D1 to 15D4 and each channel
First sample data (stored in a restart header described later) of each frame of “1” to “6” is applied to the unpacking circuit 8 and the formatting circuit 19. The sampling frequency (fs) and the number of quantization bits (Qb) when the PCM data is A / D converted are applied to the packing circuit 18 and the formatting circuit 19.
The prediction circuits 13D1, 13D2, and 15D1 to 15D4 respectively calculate the PCM data of each channel “1” to “6”.
A plurality of predictors (not shown) having different characteristics calculate a plurality of linear prediction values of the current signal from a past signal in the time domain, and then perform prediction for each predictor from the original PCM data and the plurality of linear prediction values. Calculate the residual. The following buffer
The selectors 14D1, 14D2, 16D1 to 16D4 are prediction circuits 13D1, 13D2, 15D1 to 15D, respectively.
4 is temporarily stored, and the minimum value of the prediction residual is selected for each subframe specified by the selection signal / DTS (decoding time stamp) generator 17.

The selection signal / DTS generator 17 applies the bit number flag of the prediction residual to the packing circuit 18 and the formatting circuit 19, and outputs a predictor selection flag indicating the predictor having the minimum prediction residual. To the formatting circuit 19, the correlation coefficient a in the equation (1) and the DTS indicating the time at which the decoding side extracts the stream data from the input buffer 22a (FIG. 7). Packing circuit 18
Are buffer / selectors 14D1, 14D2, 16D1-1.
The prediction residual for 6 ch selected by 6D4 is packed with the specified bit number based on the bit number flag specified by the selection signal / DTS generator 17. In the PTS generator 17c, the decoding side is the output buffer 110 (FIG. 7).
A PTS (Presentation Time Stamp) indicating the time at which the PCM data is to be extracted from the PCM is generated and output to the formatting circuit 19.

The following formatting circuit 19 is shown in FIGS.
Format as user data as shown in FIG.
Is a variable-rate bit stream (sub-stream) BS0 including 2ch “1” and “2” prediction coded data related to the forward group.
And a variable-rate bit stream (substream) BS1 including 4ch “3” to “6” prediction coded data relating to other groups, and a bitstream header (restart header) provided before substreams BS0 and BS1 ). Also, substream B
One frame of S0 and BS1 includes: a frame header; first sample data of one frame of each channel “1” to “6”; and selection of a predictor for each subframe of each channel “1” to “6”. A flag; a bit number flag for each subframe of each ch “1” to “6”; a prediction residual data string (variable bit number) for each ch “1” to “6”; Are multiplexed. According to such predictive coding, the original signal has, for example, a sampling frequency (fs) = 96 kHz.
In the case of z, the number of quantization bits (Qb) = 24 bits, and 6 channels, a compression ratio of 71% can be realized.

When the variable rate bit stream data predictively encoded by the encoding units 2'-1 and 2'-2 shown in FIG. 2 is recorded on a DVD audio disc as an example of a recording medium, FIG. The audio (A) pack shown is packed. This pack has 4 bytes of pack start information and 6 bytes of SCR (Syst) for 2034 bytes of user data (A packet, V packet).
em Clock Reference (system time reference value) information, a 3-byte Mux rate (rate) information, and a 1-byte stuffing that add a pack header of a total of 14 bytes (1 pack = 2048 bytes in total) . In this case, the time stamp SCR information is
In the first pack, the time of the A-pack in the same title can be managed by setting it as “1” and continuing within the same title.

As shown in detail in FIG. 5, the A packet of the compressed PCM has a packet header of 9 to 22 bytes and a compressed P packet.
The CM private header and 1 to 2015 bytes of audio data (compressed P
CM). And DTS and PTS
5 is set in the packet header of FIG. 5 (specifically, the PTS is set in the 10th to 14th bytes and the DTS is set in the 15th to 19th bytes). The private header of the compressed PCM is: 1-byte substream ID, 2 bytes of UPC / EAN-ISRC (Universal Prism).
oduct Code / European Article Number-International S
tandard Recording Code) number and UPC / EAN-
ISRC data, 1-byte private header length, 2-byte first access unit pointer, 4-byte audio data information (ADI), and 0-7 byte stuffing bytes. ing.

Then, both the forward access unit search pointer for searching for the access unit one second later and the backward access unit search pointer for searching for the access unit one second earlier in the ADI are one.
Set in bytes. Specifically, the forward access unit search pointer is set in the first byte of the ADI, and the backward access unit search pointer is set in the eighth byte. Thus, ADI can store up to 2015 bytes of audio data in order to reduce it to 4 bytes in compressed PCM.

The audio data area in the compressed PCM (PPCM) audio packet shown in FIG. 5 is composed of a plurality of PPCM access units as shown in FIG. 6, and the PPCM access unit is composed of PPCM sync information and subpackets. I have. First PP
The subpacket in the CM access unit includes a directory, a substream “BS0”, a CRC (1 byte or 2 bytes), a substream “BS1”,
C and extra information, and the sub-streams “BS0” and “BS1” are composed of only PPCM blocks. Sub-packets in the second and subsequent PPCM access units also include a directory, a sub-stream “BS0”, a CRC, and a sub-stream “BS1”.
, CRC and extra information, and the sub-streams “BS0” and “BS1” have a restart header and P
It is composed of PCM blocks. The extra information has at least a size adjustment function. That is, when the incoming data has a fixed rate (CBR), the number of samples per packet is 40, 80, 1 depending on the sampling frequency fs as described above.
60, the data length per packet may not match the size of the subpacket depending on the determined number of samplings. In order to match it with the size of the subpacket, for example, ,
The size is adjusted by adding 0, 0, etc. In addition, text data or the like can be used as the data for adjusting the size.

The PPCM sync information (hereinafter also referred to as synchronization information) includes the following information. -Number of samples per packet: 40, 80 or 160 is selected according to the sampling frequency fs. If the data rate is VBR, "0" (an identifier indicating that the data in the subpacket is compressed data of VBR); if the data rate is CBR, "1" (the data in the subpacket has a fixed rate) Identifier indicating that sampling frequency fs and number of quantization bits Qb Channel assignment information

Next, referring to FIG. 7, the decoding units 3'-1,
3′-2 will be described. The variable rate bit stream data BS0 and BS1 in the above format are separated by the deformatting circuit 21. And each channel
The head sample data of one frame of “1” to “6” and the predictor selection flag are stored in the prediction circuits 24D1 and 24D, respectively.
2, 23D1 to 23D4, and each channel “1” to
The bit number flag of “6” is applied to the unpacking circuit 22. The SCR, the DTS, and the prediction residual data string are applied to the input buffer 22a, and the PTS is applied to the output buffer 110. An identifier indicating whether the data rate is VBR or CBR is used as each of the predictors 24D1, 24D2,
23D1, 23D2, 23D3, 23D4,
In these, a processing program for input / output data corresponding to the identifier is determined and processed. In the case of VBR, it is necessary to switch the processing program and to load the input data every time, so that it takes time for processing. However, in the case of CBR, the processing rate does not need to be changed because the processing rate is fixed because the rate is fixed. Be faster. The sampling frequency fs and the number of quantization bits Qb are applied to the D / A converter 102. Here, a plurality of predictors (not shown) in the prediction circuits 24D1, 24D2, and 23D1 to 23D4 are each a prediction circuit 1 on the encoding side.
The same characteristics as those of the plurality of predictors in 3D1, 13D2, and 15D1 to 15D4, and those having the same characteristics are selected by the predictor selection flag.

An audio packet is first separated from the audio pack by the reformatting circuit 21. Next, stream data (predicted residual data string) is separated from the audio packet, and the bit streams BS0 and BS0 are separated.
1 is taken out. Further, the SCR is taken out and taken into the input buffer 22a for each access unit according to the timing by the SCR as shown in FIG. Here, the data amount of one access unit is
For example, when fs = 96 kHz (1/96 kHz)
The second is a variable length as shown in FIGS. 9 and 10A in detail. Then, the stream data stored in the input buffer 22a is read out by the FIFO based on the DTS and applied to the unpacking circuit 22.

The unpacking circuit 22 is provided for each channel "1" to
The prediction residual data string of “6” is separated based on each bit number flag, and is divided into prediction circuits 24D1, 24D2, and 23, respectively.
It outputs to D1-23D4. Prediction circuits 24D1, 24D
2, 23D1 to 23D4, the current prediction residual data of each of the channels “1” to “6” from the unpacking circuit 22 and each of the plurality of internal predictors selected by the predictor selection flag. The previous predicted value predicted by one frame is added to calculate the current predicted value, and then the PC of each sample is determined based on the first sample data of one frame.
M data is calculated and stored in the output buffer 110. The PCM data stored in the output buffer 110 is P
The data is read out and output based on the TS. Therefore, the variable-length access unit shown in FIG. 10A is expanded, and a fixed-length presentation unit shown in FIG. 10B is output.

Further, based on the sampling frequency fs and the number of quantization bits Qb in the PPCM sync information, the PC
The M data is converted by the D / A converter 102 into an analog signal. At the same time, C
When the BR identifier is detected, the position of the extra data in the directory is detected, and further, for example, data of 0, 0... Or extra data for size adjustment such as text data is detected, it is converted to text data. In some cases, the extra data is supplied from the unpacking circuit 22 to a text data decoding circuit (not shown), where the data is decoded, extracted as text data, and output through the output buffer 110. On the other hand, if the extra data is 0,0... Data, no processing is performed. If the text data decoder circuit is not prepared, this processing is passed. Also, where
When the search reproduction is instructed via the operation unit 101, the control unit 100 performs the search based on the forward access unit search pointer (one second ahead) and the backward access unit search pointer (one second before) shown in FIG. Play the access unit. As this search pointer, one second ahead,
Instead of one second before, two seconds before and two seconds before may be used.

When variable-rate bit stream data predictively coded by the coding units 2'-1 and 2'-2 shown in FIG. 2 is transmitted through a network, the coding side performs the processing shown in FIG. (Step S41), add a packet header (step S42), and send this packet out onto the network (step S43).

On the decoding side, as shown in FIG. 12A, the header is removed (step S51), the data is restored (step S52), and the data is stored in the memory to wait for decoding (step S53). . Then, when decoding is performed, as shown in FIG. 12B, deformatting is performed (step S61), and then the input buffer 22
The input / output control of a is performed (step S62), and then the unpacking is performed (step S63). At this time, if there is a search reproduction instruction, the search pointer is decoded. Next, a predictor is selected and decoded based on the flag (step S64), input / output control of the output buffer 110 is performed (step S65), and the original multi-channel is restored (step S66). This is output (step S67), and thereafter, this is repeated.

In the above embodiment, 2ch "1" and "2" relating to the front group are converted by "1" = Lf + Rf "2" = Lf-Rf and are predictively coded. , Down-mixing the multi-channels to generate stereo 2-ch data (L, R), and then the following equation (1) ′ “1” = L + R “2” = LR “3” to “5” are the same as “6”. = Lfe-C (1) ′ and may be subjected to predictive coding (second
Embodiment). In this case, the mix & matrix circuit 4 ′ on the decoding side adds the channels “1” and “2” to the channel L, and subtracts the channel R by adding the channels “1” and “2”.
Can be generated.

As a third embodiment, as shown in FIG. 13, instead of 2ch "1" and "2", multi-channels are downmixed by equation (2) and stereo 2ch data (L, R) is obtained. The stereo 2ch (L, R) and the 4ch “3” to “6” may be generated and predictively coded. In the second and third embodiments, since the front left (Lf) and the front right (Rf) are not transmitted to the decoding side, the decoding side generates them according to equations (1) and (2).

Next, a fourth embodiment will be described with reference to FIG. 14, FIG. 15, and FIG. In the above embodiment,
Although one group of correlated signals “1” to “6” are configured to be predictively coded, in the fourth embodiment, a plurality of groups of correlated signals are generated and predicted and coded. It is configured to select the prediction coded data of the group having the highest compression ratio. Further, in this embodiment, the encoding in one group is not classified and converted into 2ch for the front group and 4ch for the other group as in each of the above-described embodiments. FIG. 1 shows a configuration in which the combined encoding process is performed.
4 is shown as a diagram corresponding to FIG. Also,
FIG. 15 shows a detailed block of the encoding unit.
In this embodiment, n correlation circuits 1-1 to 1-n are provided on the mix & matrix circuit 1 'side.
These n correlation circuits 1-1 to 1-n are, for example, 6 ch (L
f, C, Rf, Ls, Rs, Lfe) are converted into n types of 6-channel signals “1” to “6” having different correlations.

For example, the first correlation circuit 1-1 converts as follows: "1" = Lf "2" = C- (Ls + Rs) / 2 "3" = Rf-Lf "4" = Ls-a × Lfe “5” = Rs−b × Rf “6” = Lfe Further, the n-th correlation circuit 1-n performs conversion as follows. “1” = Lf + Rf “2” = C−Lf “3” = Rf−Lf “4” = Ls−Lf “5” = Rs−Lf “6” = Lfe−C

A prediction circuit 15 and a buffer / selector 16 are provided for each of the correlation circuits 1-1 to 1-n, and the group having the highest compression ratio is determined based on the data amount of the minimum prediction residual for each group. Are selected by the correlation selection signal generator 17b. At this time, the formatting circuit 19 adds and multiplexes the selection flag (correlation circuit selection flag, correlation coefficients a and b of the correlation circuit).

FIG. 16 shows a data area corresponding to FIG. 6 described above. In this embodiment, the sub-stream "B"
Instead of using “S1”, the sub-stream “BS0” alone is used.

On the decoding side shown in FIG. 17, n correlator circuits 4 are provided for the correlator circuits 1-1 to 1-n on the encoding side.
−1 to 4-n (or one correlation circuit (not shown) whose coefficients a and b can be changed) are provided. Note that n shown in FIG.
When the prediction circuits of the groups have the same configuration, the decoding device does not need to provide the prediction circuits of n groups as shown in FIG. 17, and may use the prediction circuits of one group. Then, one of the correlation circuits 4-1 to 4-n is selected based on the selection flag transmitted from the encoding device, or the coefficients a and b are selected.
And set the original 6 ch (Lf, C, Rf, Ls, Rs, L
fe), and down-mixes the multi-channels according to equation (2) to generate stereo 2-ch data (L, R).

In the first embodiment, one kind of correlation signal "1" to "6" is configured to be predictively coded, but the signals "1" to "6" are encoded. And the group of the original signals (Lf, C, Rf, Ls, Rs, Lfe) may be predictively coded and the group with the higher compression ratio may be selected.

[0035]

As described above, according to the present invention, a subpacket including compressed data and a data structure having a synchronization information section including its sampling frequency and the number of quantization bits are formatted. When a multi-channel audio signal is encoded with a variable compression ratio, decoding efficiency on the reproduction side can be improved.

[Brief description of the drawings]

FIG. 1 is a block diagram illustrating a first embodiment of a speech encoding device and a speech decoding device according to the present invention.

FIG. 2 is a block diagram illustrating an encoding unit of FIG. 1 in detail.

FIG. 3 is an explanatory diagram showing a bit stream encoded by an encoding unit shown in FIGS. 1 and 2;

FIG. 4 is an explanatory diagram showing a format of a DVD pack.

FIG. 5 is an explanatory diagram showing a format of a DVD audio pack.

6 is an explanatory diagram showing the format of the audio data area in FIG. 5 in detail.

FIG. 7 is a block diagram illustrating a decoding unit of FIG. 1 in detail.

FIG. 8 is a timing chart showing write / read timings of the input buffer of FIG. 7;

FIG. 9 is an explanatory diagram showing the amount of compressed data for each access unit.

FIG. 10 is an explanatory diagram showing an access unit and a presentation unit.

FIG. 11 is a flowchart showing a voice transmission method.

FIG. 12 is a flowchart showing a voice transmission method.

FIG. 13 is a block diagram illustrating a speech encoding device and a speech decoding device according to a third embodiment.

FIG. 14 is a block diagram illustrating a fourth embodiment of the speech encoding device and the speech decoding device according to the present invention.

FIG. 15 is a block diagram illustrating a speech encoding device according to a fourth embodiment.

FIG. 16 is an explanatory diagram of another embodiment corresponding to FIG. 6;

FIG. 17 is a block diagram illustrating a speech decoding device according to a fourth embodiment.

[Explanation of symbols]

1 '6ch Mix & Matrix Circuit 13D1, 13D2, 15D1-15D4 Prediction Circuit (Buffer / Selector 14D1, 14D2, 16D1-1
A compression means is constituted together with 6D4. 14D1, 14D2, 16D1-16D4 buffer
Selector 17 Selection signal / DTS generator (timing generating means) 17c PTS generator (timing generating means) 19 Formatting circuit (Formatting means) 21 Deformatting circuit (Separating means) 22 Unpacking circuit 22a Input buffer 24D1, 24D2, 23D1 to 23D4 Prediction circuit (expansion means) 100 Control unit 102 D / A converter 110 Output buffer

Claims (9)

[Claims] 1. A compression means for compressing a multi-channel audio signal having a certain sampling frequency and the number of quantization bits by a predictive coding method for each channel, and a subpacket including data for each channel compressed by the compression means. Means for formatting into a data structure having a synchronization information section including the sampling frequency and the number of quantization bits. 2. The audio coding apparatus according to claim 1, wherein the data structure is formatted by adding a header including SCR information corresponding to a sub-packet and a synchronization information section. . 3. A multi-channel audio signal having a certain sampling frequency and the number of quantization bits is compressed by a predictive coding method for each channel. A recording medium which is formatted and recorded in a data structure having a synchronization information section including the number of encoded bits. 4. The recording medium according to claim 3, wherein the data structure is formatted by adding a header including SCR information corresponding to a sub-packet and a synchronization information part. 5. A multi-channel audio signal having a certain sampling frequency and the number of quantization bits is compressed by a predictive coding method for each channel, a subpacket including the compressed data for each channel, a sampling frequency and a quantization What is claimed is: 1. An audio decoding apparatus for decoding a data structure having a synchronization information section including a number of encoded bits, comprising: means for separating the data structure into a subpacket and a synchronization information section; and decompressing compressed data in the subpacket for each channel. And a means for converting the expanded audio data into an analog audio signal based on the sampling frequency and the number of quantization bits in the synchronization information section. 6. A multi-channel audio signal having a certain sampling frequency and quantization bit number is compressed by a predictive coding method for each channel, and a sub-packet including the compressed data for each channel, a sampling frequency and a quantization value An audio decoding apparatus for decoding a data structure having a synchronization information part including a number of encoded bits and a header including SCR information corresponding to the subpacket and the synchronization information part, wherein the SCR information included in the header is separated. A first separating unit, a buffer for holding the subpacket and the synchronization information section based on the separated SCR information, and a second for separating the subpacket and the synchronization information section held in the buffer. And decompressing the compressed data in the subpacket for each channel based on the identifier in the synchronization information section. A long section, and means for converting the analog audio signal based on the decompressed audio data to a sampling frequency and number of quantization bits in the synchronization information unit includes speech decoding apparatus. 7. A multi-channel audio signal having a certain sampling frequency and a quantized bit number is compressed by a predictive coding method for each channel, a subpacket including the compressed data for each channel, and a sampling frequency and quantization A voice transmission method characterized by transmitting, via a communication line, a packet having a data structure formatted into a data structure having a synchronization information section including a number of encoded bits. 8. The audio transmission method according to claim 7, wherein the data structure is formatted by adding a header including SCR information corresponding to a subpacket and a synchronization information part. 9. A multi-channel audio signal having a certain sampling frequency and the number of quantization bits is compressed by a predictive coding method for each channel, a subpacket including the compressed data for each channel, a sampling frequency and a quantization A transmission medium for transmitting a packet having a data structure formatted into a data structure having a synchronization information section including a number of encoded bits.