JP2009145919A - Audio coding method, audio decoding method, and audio signal transmission method - Google Patents

Abstract

<P>PROBLEM TO BE SOLVED: To manage processing time on a reproduction side, when a multichannel audio signal is to be coded at a variable compression rate. <P>SOLUTION: Prediction circuits 13D1, 13D2, and 15D1-15D4, and buffer selection units 14D1, 14D2, and 16D1-16D4 predictively encode six-channel audio signals. A decoding time stamp (DTS) generating unit 17 generates decoding time stamp information, indicating the timing for reading compressed data in an input buffer 22a on a decoding side, according to the prediction coding data amount for each channel. A formatting circuit 19 performs formatting to create a data structure, having a packet header containing the decoding time stamp information and a packet containing user data containing the compression data. <P>COPYRIGHT: (C)2009,JPO&INPIT

Description

  The present invention relates to a speech encoding method, speech decoding method, and speech signal transmission method for compressing a multi-channel speech signal with variable length.

As a method of compressing an audio signal with a variable length, the present inventor has proposed an earlier application (Japanese Patent Application No. 9-28915).
9), a plurality of linear prediction values of the current signal are calculated from a past signal in the time domain by a plurality of predictors having different characteristics with respect to one channel of the original digital audio signal, A prediction encoding method is proposed in which a prediction residual for each predictor is calculated from a plurality of linear prediction values, and a minimum value of the prediction residual is selected.

In the above method, a certain degree of compression effect can be obtained when the original digital audio signal has a sampling frequency = 96 kHz and the number of quantization bits = 20 bits.
In the D audio disc, this doubled sampling frequency (= 192 kHz) is used, and the number of quantization bits tends to be 24 bits, so it is necessary to improve the compression rate. In addition, the sampling frequency and the number of quantization bits in multichannel may be different for each channel.

By the way, since a compression method such as the predictive coding method has a variable compression rate (VBR: variable bit rate), when a multi-channel audio signal is predictively encoded, the amount of data for each channel greatly changes over time. To do. When such data is transmitted, it is transmitted as a data stream instead of parallel for each channel.

Therefore, in order to enable reproduction (presentation) of such a variable length data stream in synchronization with each channel on the reproduction side (decoding side), the data stream stored in the input buffer is read and output to the decoder. Therefore, it is necessary to manage the decoding time indicating the timing for the output and the reproduction time indicating the timing for reading the decoded data accumulated in the output buffer and outputting (presenting) the data to a speaker or the like. Further, the playback side must manage the time for searching and playing back such a variable length data stream.

Therefore, the present invention provides an audio encoding method, an audio decoding method, and an audio signal transmission method capable of managing the processing time on the playback side when encoding a multi-channel audio signal with a variable compression rate. Objective.

In order to achieve the above object, the present invention comprises the following means 1) to 3).
That is,
1) Downmixing the original multi-channel audio signal and converting it to a stereo 2-channel audio signal;
Transforming the original multi-channel into a number of correlated channels that are correlated channels, the number of channels being less than the two channels by a predetermined matrix operation;
The audio signals of the stereo 2 channel and the correlation channel are obtained for each channel in response to the input audio signal in a frame unit for a predetermined time, and a plurality of linear prediction methods having different characteristics are used to obtain a time sample by a plurality of linear prediction methods. The frame is further divided by the linear prediction method in which the linear prediction value of the current signal is predicted from the past and the prediction residual obtained from the predicted linear prediction value and the speech signal is minimized. Selecting and encoding in subframe units;
The data structure includes header information and user data including a compressed PCM access unit, and the compressed PCM access unit is provided for each subframe, and each channel for each subframe selected in the step is selected. Predictive coded data including a linear prediction method and a prediction residual are grouped into bit streams grouped into a first group of stereo 2 channels and a second group of correlation channels arranged in the compressed PCM access unit. If the compressed PCM access unit is the first one in the frame, a restart header is provided in the bitstream and the first sample value is stored, and each compressed PCM is further stored. Data in the bitstream is allowed in the access unit Comprising: providing synchronization information part including a VBR identifier indicating that the compressed data bit rate compression,
A speech encoding method comprising:
2) A speech decoding method for decoding an original speech signal from data encoded by the speech encoding method according to claim 1,
Extracting a compressed PCM access unit from the user data;
Extracting a VBR identifier from the compressed PCM access unit;
Based on the extracted identifier, the head sample value is extracted from the compressed PCM access unit having the restart header, and a prediction residual and linear prediction are obtained from a grouped bit stream arranged in each compressed PCM access unit. Retrieving predictive encoded data in units of subframes including a method;
Calculating a prediction value based on the first sample value and the prediction residual and linear prediction method;
Restoring the stereo 2-channel audio signal and the original multi-channel audio signal from the calculated predicted value;
A speech decoding method comprising:
3) Downmixing the original multi-channel audio signal to convert it into a stereo 2-channel audio signal;
Transforming the original multi-channel into a number of correlated channels that are correlated channels, the number of channels being less than the two channels by a predetermined matrix operation;
The audio signals of the stereo 2 channel and the correlation channel are obtained for each channel in response to the input audio signal in a frame unit for a predetermined time, and a plurality of linear prediction methods having different characteristics are used to obtain a time sample by a plurality of linear prediction methods. The frame is further divided by the linear prediction method in which the linear prediction value of the current signal is predicted from the past and the prediction residual obtained from the predicted linear prediction value and the speech signal is minimized. Selecting and encoding in subframe units;
A data structure including a packet header and user data including a compressed PCM access unit is provided, and the compressed PCM access unit is provided for each subframe, and each channel for each subframe selected in the step is selected. Predictive coded data including a linear prediction method and a prediction residual are grouped into bit streams grouped into a first group of stereo 2 channels and a second group of correlation channels arranged in the compressed PCM access unit. The compressed PCM access is provided with a synchronization information section that stores the number of samples according to the sampling frequency of the audio signal, and further includes a VBR identifier indicating that the data in the bitstream is compressed data with variable bit rate compression. The unit is the first one in the frame If that is further coded by the speech coding method comprising the step of storing said first sample value provided with the restart header to the bit stream,
An audio signal transmission method comprising: packetizing and transmitting predictive encoded data including the selected head sample value, a prediction residual, and a linear prediction method.

As described above, according to the present invention, the sub packet including the compressed data and the data structure including the synchronization information portion including the channel allocation information of the data are formatted, so that, for example, on the playback side, the channel It is possible to provide a speech encoding method that can perform accurate data processing according to the number.

It is a block diagram which shows 1st Embodiment of the audio | voice coding apparatus with which this invention is applied, and the audio | voice decoding apparatus corresponding to it. It is a block diagram which shows the encoding part of FIG. 1 in detail. It is explanatory drawing which shows the bit stream encoded by the encoding part of FIG. 1, FIG. It is explanatory drawing which shows the format of the pack of DVD. It is explanatory drawing which shows the format of the audio pack of DVD. It is a block diagram which shows the decoding part of FIG. 1 in detail. 7 is a timing chart showing write / read timings of the input buffer of FIG. 6. It is explanatory drawing which shows the compressed data amount for every access unit. It is explanatory drawing which shows an access unit and a presentation unit. It is a flowchart which shows the audio | voice transmission method. It is a flowchart which shows the audio | voice transmission method. It is a block diagram which shows 3rd Embodiment of the audio | voice encoding apparatus with which this invention is applied, and the audio | voice decoding apparatus corresponding to it. It is a block diagram which shows the audio | voice coding apparatus of 4th Embodiment. It is a block diagram which shows the audio | voice decoding apparatus of 4th Embodiment.

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 voice encoding apparatus to which the present invention is applied and a corresponding voice decoding apparatus. FIG.
FIG. 3 is an explanatory diagram showing a bit stream encoded by the encoding unit shown in FIGS. 1 and 2, FIG. 4 is an explanatory diagram showing a format of a DVD pack, and FIG.
Is an explanatory diagram showing the format of a DVD audio pack, FIG. 6 is a block diagram showing in detail the decoding unit of FIG. 1, FIG. 7 is a timing chart showing write / read timing of the input buffer of FIG. 6, and FIG. FIG. 9 is an explanatory diagram showing an access unit and a presentation unit.

Here, as the multi-channel method, for example, the following four methods are known.
(1) 4-channel system Like the Dolby Surround system, a total of 4 channels of 3 channels for the front L, C, and R + 1 channel for the rear S (2) 5 channels system Like no Dolby AC-3 system SW channel , Forward L
, C, R 3 channels + rear SL, SR 2 channels in total 5 channels (3) 6 channel system DTS (Digital Theater System) system and Dolby AC-
6 channels (L, C, R, SW (Lfe), SL, SR)
(4) 8-channel system Like the SDDS (Sony Dynamic Digital Sound) system, a total of 8 channels including 6 channels of forward L, LC, C, RC, R, and SW + 2 channels of backward SL and SR The 6-channel (ch) mix & matrix circuit 1 ′ on the control side includes a front left (Lf), a center (C), a front right (Rf), an example of a multi-channel signal,
Surround Left (Ls), Surround Right (Rs), and Lfe (Low Frequency Effe
ct) 6ch PCM data 2ch "1" and "2" for the front group according to the following equation (1)
4ch “3” to “6” related to the other groups and conversion, 2ch “1” and “2” are converted 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
However, 0 ≦ a ≦ 1 (1)
As shown in detail in FIG. 2, the first and second encoding units 2′-1, 2′-2 constituting the encoding unit 2 ′ are respectively 2ch “1”, “2” and 4ch “3” to “3” to “3” to “3”. The PCM data of “6” is predictively encoded, and the predictive encoded data is transmitted to the decoding side via the recording medium 5 and the communication medium 6 in a bit stream as shown in FIG. On the decoding side, the first and second decoding units 3 ′ constituting the decoding unit 3 ′
-1, 3′-2, 2ch “1” relating to the front group, as shown in detail in FIG.
”,“ 2 ”and 4ch“ 3 ”to“ 6 ”predictive encoded data relating to other groups are decoded into PCM data.

Next, the original 6ch (Lf, C, R) based on the formula (1) by the mix & matrix circuit 4 ′
f, Ls, Rs, Lfe) are restored, and the original 6ch and the coefficient mij (i = 1, 2) are restored.
, J = 1, 2 to 6), stereo 2ch data (L, R) is generated as in the following equation (2).

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)
The encoding units 2′-1 and 2′-2 will be described in detail with reference to FIG. Each channel “1” to “
6 "PCM data is stored in one frame buffer 10 for each frame. And 1
The sample data of each channel “1” to “6” of the frame is the prediction circuits 13D1 and 13D2, respectively.
, 15D1 to 15D4 and the top sample data of each frame “1” to “6” is applied to the formatting circuit 19. Prediction circuits 13D1, 13D2, 1
5D1 to 15D4 respectively represent a plurality of linear prediction values of a current signal from a past signal in a time domain by a plurality of predictors (not shown) having different characteristics with respect to PCM data of each channel “1” to “6”. Then, a prediction residual for each predictor is calculated from the original PCM data and the plurality of linear prediction values. The subsequent buffer / selectors 14D1, 14D2, 16D1 to 16D4 temporarily store the prediction residuals calculated by the prediction circuits 13D1, 13D2, and 15D1 to 15D4, respectively, and select signals / DTS (decoding time stamp). The minimum value of the prediction residual is selected for each subframe designated by the generator 17.

The selection signal / DTS generator 17 applies a bit number flag of the prediction residual to the packing circuit 18 and the formatting circuit 19, and also includes a predictor selection flag indicating a predictor having the smallest prediction residual, and an expression ( The correlation coefficient a in 1) and the DTS indicating the time for the decoding side to extract the stream data from the input buffer 22a (FIG. 6) are applied to the formatting circuit 19. The packing circuit 18 packs the prediction residuals for 6ch selected by the buffers / selectors 14D1, 14D2, 16D1 to 16D4 with the designated number of bits based on the bit number flag designated by the selection signal / DTS generator 17. . The PTS generator 17c generates a PTS (Presentation Time Stamp) indicating the time when the decoding side takes out the PCM data from the output buffer 110 (FIG. 6), and outputs the PTS (Presentation Time Stamp) to the formatting circuit 19.

The subsequent formatting circuit 19 formats the user data as shown in FIGS. The user data (subpacket) shown in FIG. 3 includes 2ch “1” related to the front group,
Variable rate bitstream (substream) BS0 including predictive encoded data of “2”
A variable-rate bitstream (substream) BS1 including 4ch “3” to “6” predictive encoded data related to other groups, and a bitstream header (restart header) provided before the substreams BS0 and BS1 ).
Also, one frame of substream BS0, BS1 is a frame header,
・ First sample data of one frame of each channel “1” to “6”,
A predictor selection flag for each subframe of each channel “1” to “6”;
A bit number flag for each subframe of each channel “1” to “6”;
-Predictive residual data string (number of variable bits) of each ch "1" to "6",
・ Ch “6” coefficient a
Are multiplexed. According to such predictive coding, when the original signal is, for example, sampling frequency = 96 kHz, quantization bit number = 24 bits, and 6 channels, a compression rate of 71% can be realized.

When the variable rate bit stream data predictively encoded by the encoding units 2′-1, 2′-2 shown in FIG. 2 is recorded on a DVD audio disk as an example of a recording medium,
It is packed in an audio (A) pack shown in FIG. This pack has 2034 bytes of user data (A packet, V packet), 4 bytes of pack start information, 6
Byte (SCR) (System Clock Reference) information, 3-byte Mux rate information, and 1-byte stuffing total 14-byte pack header are added (1 pack = 2048 bytes in total). In this case, the time of the A pack in the same title can be managed by setting the SCR information as a time stamp as “1” in the first pack and continuing in the same title.

As shown in detail in FIG. 5, the compressed PCM A packet includes a 19- or 14-byte packet header, a compressed PCM private header, and 1 to 201 in the format shown in FIG.
It is composed of 1-byte audio data (compressed PCM). And DTS and P
TS is included in the packet header of FIG. 5 (specifically, P at the 10th to 14th bytes of the packet header).
TS is set in the 15th to 19th bytes). The compressed PCM private header is
A 1-byte substream ID,
・ 2-byte UPC / EAN-ISRC (Universal Product Code / European Article Nu
mber-International Standard Recording Code) number and UPC / EAN-I
SRC data,
-1 byte private header length,
A 2-byte first access unit pointer;
8 bytes of audio data information (ADI)
・ With stuffing byte of 0-7 bytes,
It is made up of. Both the forward access unit search pointer for searching the access unit after 1 second in the ADI and the backward access unit search pointer for searching for the access unit before 1 second are both 1 byte (specifically, Is ADI 7
The front access unit search pointer is set in the byte and the backward access unit search pointer is set in the eighth byte.

Next, the decoding units 3′-1 and 3′-2 will be described with reference to FIG. The variable rate bit stream data BS0 and BS1 in the above format are separated by the deformatting circuit 21. Then, the first sample data and predictor selection flag of one frame of each channel “1” to “6” are applied to the prediction circuits 24D1, 24D2, and 23D1 to 23D4, respectively.
The bit number flags of each channel “1” to “6” are applied to the unpacking circuit 22. Also,
The SCR, DTS, and prediction residual data string are applied to the input buffer 22a, and PTS is applied to the output buffer 110. Here, the prediction circuits 24D1, 24D2, 23D1 to 23D4
Each of the plurality of predictors (not shown) includes prediction circuits 13D1, 13D2, 15 on the encoding side.
The characteristics are the same as those of the plurality of predictors in D1 to 15D4, and the same characteristics are selected by the predictor selection flag.

The stream data (predicted residual data string) separated by the deformatting circuit 21 is taken and stored in the input buffer 22a for each access unit by the SCR as shown in FIG. Here, the data amount of one access unit is, for example, fs = 96 kHz.
In the case of (1) / (96 kHz) seconds, the length is variable as shown in detail in FIGS. 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 separates the prediction residual data strings of the channels “1” to “6” based on the bit number flags and outputs them to the prediction circuits 24D1, 24D2, and 23D1 to 23D4, respectively. Each of the prediction circuits 24D1, 24D2, 23D1 to 23D4 uses the current prediction residual data of each channel “1” to “6” from the unpacking circuit 22 and a predictor selection flag among a plurality of internal predictors. The previous prediction value predicted by each selected one is added to calculate the current prediction value, and then the PCM data of each sample is calculated and stored in the output buffer 110 with reference to the first sample data of one frame. Is done. Output buffer 110
The PCM data stored in is read and output based on the PTS. Therefore, the variable-length access unit shown in FIG. 9A is expanded, and the fixed-length presentation unit shown in FIG. 9B is output.

When search reproduction is instructed via the operation unit 101, the front access unit search pointer indicating 1 second ahead placed in the ADI shown in FIG.
The access unit is reproduced based on the backward access unit search pointer indicating the second after. This search pointer may be one second ahead and two seconds ahead instead of one second ahead and one second ahead.

When the variable rate bit stream data predictively encoded by the encoding units 2′-1 and 2′-2 shown in FIG. 2 is transmitted via the network, the encoding side uses the transmission unit as shown in FIG. (Step S41), then a packet header is added (step S42), and then the packet is sent out on the network (step S43).

As shown in FIG. 11A, the decoding side removes the header (step S51), then restores the data (step S52), then stores this data in the memory and waits for decoding (step S53). When decoding is performed, as shown in FIG. 11B, deformatting is performed (step S61), input / output control of the input buffer 22a is performed (step S62), and then unpacking is performed (step S61). S63). At this time, if there is a search reproduction instruction, the search pointer is decoded. Next, the predictor is selected and decoded based on the flag (step S64), then the input / output control of the output buffer 110 is performed (step S65), and then the original multi-channel is restored (step S66). This is repeated (step S67).

In the above embodiment, 2ch “1” and “2” related to the front group are set to “1” = Lf + Rf.
“2” = Lf−Rf
However, instead, the multi-channel is downmixed according to Equation (2) to generate stereo 2ch data (L, R),
Next, the following formula (1) ′
“1” = L + R
“2” = LR
“3” to “5” are the same. “6” = Lfe-C (1) ′
(2nd embodiment). In this case, the decoding-side mix & matrix circuit 4 ′ can generate channel R by adding channels “1” and “2” and subtracting channel L by subtraction.

Further, as shown in FIG. 12 as the third embodiment, instead of 2ch “1” and “2”, the formula (
The stereo 2ch data (L, R) is generated by downmixing the multi-channel according to 2), and the stereo 2ch (L, R) and 4ch “3” to “6” may be predictively encoded. . In the second and third embodiments, the front left (Lf) and the front right (Rf)
Is not transmitted to the decoding side, the decoding side generates this according to equations (1) and (2).

Next, a fourth embodiment will be described with reference to FIGS. The above embodiment is configured to predictively encode one group of correlated signals “1” to “6”.
In the fourth embodiment, a plurality of groups of correlated signals are generated and subjected to predictive encoding, and the predictive encoded data of the group having the highest compression rate is selected. For this reason, the encoding unit shown in FIG. 13 is provided with first to n-th correlation circuits 1-1 to 1-n, and these n correlation circuits 1-1 to 1-n are, for example, 6ch (Lf, C , Rf, Ls, Rs, Lfe) PCM
The data is converted into n types of 6ch 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
The n-th correlation circuit 1-n converts as follows.

“1” = Lf + Rf
“2” = C−Lf
“3” = Rf−Lf
“4” = Ls−Lf
“5” = Rs−Lf
“6” = Lfe-C
Further, a prediction circuit 15 and a buffer / selector 16 are provided for each of the correlation circuits 1-1 to 1-n.
The group having the highest compression rate is selected by the correlation selection signal generator 17b based on the data amount of the minimum value of the prediction residual for each group. 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).

Further, on the decoding side shown in FIG. 14, n correlation circuits 4-1 to 4-n (or coefficients a and b can be changed with respect to the correlation circuits 1-1 to 1-n on the encoding side. One correlation circuit (omitted) is provided. When the n groups of prediction circuits shown in FIG. 13 have the same configuration, the decoding device does not need to have n groups of prediction circuits as shown in FIG. Then, based on the selection flag transmitted from the encoding device, the correlation circuit 4
Select one of -1 to 4-n, or set the coefficients a and b to the original 6ch (Lf, C, Rf, Ls
, Rs, Lfe), and multi-channel is downmixed according to Equation (2) to generate stereo 2ch data (L, R).

In the first embodiment described above, one type of correlation signal “1” to “6” is configured to be predictively encoded. The group of signals “1” to “6” Original signal (Lf, C,
Rf, Ls, Rs, Lfe) may be predictively encoded, and the group with the higher compression rate may be selected.

1 '6ch mix & matrix circuit 13D1, 13D2, 15D1-15D4 Prediction circuit (buffer / selector 14)
The compression means is configured together with D1, 14D2, 16D1 to 16D4. )
14D1, 14D2, 16D1 to 16D4 Buffer / selector 17 Selection signal / DTS generator (timing generation means)
17c PTS generator (timing generator)
19 Formatting circuit (formatting means)
21 Deformatting circuit (separation means)
22 Unpacking circuit 22a Input buffer 24D1, 24D2, 23D1 to 23D4 Prediction circuit (expanding means)
100 Control unit (reading means)
110 Output buffer

Claims (3)

Downmixing the original multi-channel audio signal into a stereo 2-channel audio signal;
Transforming the original multi-channel into a number of correlated channels that are correlated channels, the number of channels being less than the two channels by a predetermined matrix operation;
The audio signals of the stereo 2 channel and the correlation channel are obtained for each channel in response to the input audio signal in a frame unit for a predetermined time, and a plurality of linear prediction methods having different characteristics are used to obtain a time sample by a plurality of linear prediction methods. The frame is further divided by the linear prediction method in which the linear prediction value of the current signal is predicted from the past and the prediction residual obtained from the predicted linear prediction value and the speech signal is minimized. Selecting and encoding in subframe units;
The data structure includes header information and user data including a compressed PCM access unit, and the compressed PCM access unit is provided for each subframe, and each channel for each subframe selected in the step is selected. Predictive coded data including a linear prediction method and a prediction residual are grouped into bit streams grouped into a first group of stereo 2 channels and a second group of correlation channels arranged in the compressed PCM access unit. If the compressed PCM access unit is the first one in the frame, a restart header is provided in the bitstream and the first sample value is stored, and each compressed PCM is further stored. Data in the bitstream is allowed in the access unit Comprising: providing synchronization information part including a VBR identifier indicating that the compressed data bit rate compression,
A speech encoding method comprising: A speech decoding method for decoding an original speech signal from data encoded by the speech encoding method according to claim 1,
Extracting a compressed PCM access unit from the user data;
Extracting a VBR identifier from the compressed PCM access unit;
Based on the extracted identifier, the head sample value is extracted from the compressed PCM access unit having the restart header, and a prediction residual and linear prediction are obtained from a grouped bit stream arranged in each compressed PCM access unit. Retrieving predictive encoded data in units of subframes including a method;
Calculating a prediction value based on the first sample value and the prediction residual and linear prediction method;
Restoring the stereo 2-channel audio signal and the original multi-channel audio signal from the calculated predicted value;
A speech decoding method comprising: Downmixing the original multi-channel audio signal into a stereo 2-channel audio signal;
Transforming the original multi-channel into a number of correlated channels that are correlated channels, the number of channels being less than the two channels by a predetermined matrix operation;
The audio signals of the stereo 2 channel and the correlation channel are obtained for each channel in response to the input audio signal in a frame unit for a predetermined time, and a plurality of linear prediction methods having different characteristics are used to obtain a time sample by a plurality of linear prediction methods. The frame is further divided by the linear prediction method in which the linear prediction value of the current signal is predicted from the past and the prediction residual obtained from the predicted linear prediction value and the speech signal is minimized. Selecting and encoding in subframe units;
A data structure including a packet header and user data including a compressed PCM access unit is provided, and the compressed PCM access unit is provided for each subframe, and each channel for each subframe selected in the step is selected. Predictive coded data including a linear prediction method and a prediction residual are grouped into bit streams grouped into a first group of stereo 2 channels and a second group of correlation channels arranged in the compressed PCM access unit. The compressed PCM access is provided with a synchronization information section that stores the number of samples according to the sampling frequency of the audio signal, and further includes a VBR identifier indicating that the data in the bitstream is compressed data with variable bit rate compression. The unit is the first one in the frame If that is further coded by the speech coding method comprising the step of storing said first sample value provided with the restart header to the bit stream,
An audio signal transmission method comprising: packetizing and transmitting predictive encoded data including the selected head sample value, a prediction residual, and a linear prediction method.

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