JP3387084B2 - Recording medium, audio decoding 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 recording medium in which a multi-channel audio signal is compressed and recorded, and an audio decoding device .

[0002]

2. Description of the Related Art As a method for compressing an audio signal, the present inventor has described in 1
For the original digital audio signal of the channel, multiple predictors with different characteristics are used to calculate multiple linear prediction values of the current signal from past signals in the time domain, and the original digital audio signal and the multiple linear prediction values. We propose a predictive coding method that calculates the prediction residual for each predictor and selects the minimum value of the prediction residual.

In the above method, the original digital audio signal has a sampling frequency = 96 kHz and a quantization bit number = 2.
Although it is possible to obtain a certain degree of compression effect when the number of bits is about 0, in recent DVD audio discs, this 2
The 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 method such as a predictive coding method has a variable compression rate (VBR: variable bit rate), when predictive coding a multi-channel audio signal, The data amount of changes greatly with time. Further, when transmitting such data, it is transmitted not as parallel for each channel but as a data stream. Therefore, it is necessary for the reproducing side (decoding side) to be capable of reproducing (presenting) such a variable-length data stream in synchronization with each channel.

Therefore, the present invention is directed to a recording medium and a voice decoding device which record data capable of improving the decoding efficiency on the reproducing side when compressing and coding a multi-channel voice signal.
The purpose is to provide storage .

In order to achieve the above object, the present invention comprises the following means 1 ) to 4) . That is,

1) A multi-channel audio signal having a certain sampling frequency and a quantized bit number is obtained in response to an input audio signal for a channel as it is or for each channel correlated with each other, and a head sample value is obtained. Linear prediction values of the current signal are predicted from the past in the time domain by a plurality of linear prediction methods having different characteristics, and the prediction residual obtained from the predicted linear prediction value and the speech signal is minimized. A step of selecting a linear prediction method for predictive coding, a linear prediction method for each channel selected in the step, a subpacket for storing prediction coded data including a prediction residual and a predetermined head sample value, and reproduction A synchronization information part including a sampling frequency and a quantization bit number used when the original analog audio signal is restored on the side, Formatting into a data structure to record the formatted data in the data structure, the predictive coded data of the recorded data is a prediction value used to restore the original audio signal. Recorded as data for calculation
Recording medium characterized Tei Rukoto. 2) The sub-packet and the synchronization information part according to claim 1 and a pack header including SCR information are further formatted and recorded as one pack, and the SCR information reproduces the pack. A recording medium characterized by being used as time management information at the time. 3) A voice decoding device for decoding an original multi-channel voice signal from the data recorded on the recording medium according to claim 1, said means for separating said data structure into a sub-packet and a synchronization information part; Decompression means for decompressing the compressed data in the packet for each channel, means for converting the decompressed audio data for each channel into the original decompressed multi-channel audio signal, and the converted multi-channel audio data To an analog audio signal based on the sampling frequency and the number of quantization bits in the synchronization information section. 4) An audio decoding device for decoding an original multi-channel audio signal from the data recorded on the recording medium according to claim 2, wherein the first separating means separates the SCR information contained in the header; A buffer for holding the sub-packet and the synchronization information part based on the separated SCR information; a second separating means for separating the sub-packet and the synchronization information part held in the buffer; and the synchronization information. Decompressing means for decompressing the compressed data in the subpacket for each channel based on the identifier in the section; means for converting the decompressed audio data for each channel into an original multichannel decompressed audio signal; The converted multi-channel audio data is converted into an analog audio signal based on the sampling frequency and the number of quantization bits in the synchronization information section. A stage, having speech decoding apparatus.

[0008]

BEST MODE FOR CARRYING OUT THE INVENTION Embodiments of the present invention will be described below with reference to the drawings. 1 is a block diagram showing a first embodiment of a speech coding apparatus and speech decoding apparatus to which the present invention is applied , FIG. 2 is a block diagram showing the coding unit of FIG. 1 in detail, and FIG. 3 is FIG. 2 is an explanatory diagram showing a bitstream encoded by the encoding unit 2; FIG. 4 is an explanatory diagram showing a DVD pack format; FIG. 5 is an explanatory diagram showing a DVD audio pack format; and FIG. FIG. 7 is an explanatory diagram showing the format of the audio data area in detail, FIG. 7 is a block diagram showing the decoding section of FIG. 1 in detail, FIG. 8 is a timing chart showing the write / read timing of the input buffer of FIG. 7, and FIG. 9 is for each access unit. FIG. 10 is an explanatory diagram showing the amount of compressed data, and FIG. 10 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 method Like the Dolby Surround method, a total of 4 channels of 3 channels of front L, C, and R + 1 channel of rear S (2) 5-channel method S of Dolby AC-3 method
A total of 5 channels (3 channels of front L, C, and R + 2 channels of rear SL and SR) (3) 6-channel system DTS (Digital Theater)
System) and 6 channels (L, C, R, SW (Lfe), SL, SR) like the Dolby AC-3 system (4) 8 channel system SDDS (Sony Dynamic D
forward L, LC, C, R, like the igital Sound) method
6 channels of C, R and SW + 2 channels of rear SL and SR, total 8 channels

The 6-channel (ch) mix & matrix circuit 1'on the encoding side shown in FIG. 1 is a front left (Lf), center (C), front right (Rf), surround left ( Ls), surround light (Rs) and Lfe (Low Frequency Effe)
ct) 6ch PCM data are classified into 2ch “1” and “2” related to the front group and 4ch “3” to “6” related to other groups by the following formula (1), and converted to 2ch “1”,
"2" 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 encoding units 2'-1, 2'-2 constituting the encoding unit 2'respectively have 2ch "1", "2" and 4ch "3". ] ~ "6" PC
The M data is predictively coded for each channel, and the predictively coded data is transmitted to the decoding side via a recording medium 5 and a communication medium 6 such as a satellite line or a telephone line as a bit stream as shown in FIG. On the decoding side, the first and second decoding units 3′-1 and 3′-2, which constitute the decoding unit 3 ′, respectively, as shown in detail in FIG.
Predicted coded data of 4 channels “3” to “6” related to “2” and other groups is decoded into PCM data for each channel.

Then, based on the equation (1), the original 6ch (Lf, C, Rf, Ls,
Rs, Lfe) is restored, and stereo 2ch data (L, R) is generated by the original 6ch and coefficients mij (i = 1, 2, j = 1, 2 to 6) as shown in the following equation (2). To 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, coding units 2'-1, 2'-
2 will be described in detail. PC of each channel "1" to "6"
The M data is stored in the frame buffer 10 for each frame. Then, the sample data of each channel “1” to “6” of one frame is input to the prediction circuits 13D1 and 13D, respectively.
2, 15D1 to 15D4 applied to each channel
The leading 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 respond to the PCM data of each channel “1” to “6”,
A plurality of predictors (not shown) having different characteristics are used to calculate a plurality of linear prediction values of a current signal from a past signal in the time domain, and then original PCM data and prediction of each predictor from the plurality of linear prediction values. Calculate the residuals. Continued buffer
The selectors 14D1, 14D2, 16D1 to 16D4 are prediction circuits 13D1, 13D2, 15D1 to 15D, respectively.
The prediction residuals calculated in step 4 are temporarily stored, and the minimum value of the prediction residuals is selected for each subframe designated by the selection signal / DTS (decoding time stamp) 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 a predictor selection flag indicating a predictor with the minimum prediction residual. , The correlation coefficient a in Expression (1) and the DTS indicating the time at which the decoding side takes out the stream data from the input buffer 22a (FIG. 7) are applied to the formatting circuit 19. Packing circuit 18
Is the buffer / selector 14D1, 14D2, 16D1-1
The prediction residuals for 6 channels selected by 6D4 are packed with the specified number of bits based on the bit number flag specified by the selection signal / DTS generator 17. The decoding side of the PTS generator 17c is the output buffer 110 (FIG. 7).
PTS (Presentation Time Stamp) indicating the time to take out the PCM data is generated and output to the formatting circuit 19.

The following formatting circuit 19 is shown in FIGS.
Format the user data as shown in. Figure 3
The user data (sub-packet) shown in (1) is a variable rate bit stream (sub-stream) BS0 including predictive encoded data of 2ch “1” and “2” related to the front group.
And a variable rate bitstream (substream) BS1 including predictive coded data of 4ch "3" to "6" relating to other groups, and a bitstream header (restart header) provided before the substreams BS0 and BS1. ). Also, substream B
For one frame of S0 and BS1, a frame header, head sample data of one frame of each channel "1" to "6", and a predictor selection 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 number of bits) of each ch “1” to “6”, and a ch “6” And the coefficient a of “. According to such predictive coding, the original signal has a sampling frequency (fs) = 96 kHz, for example.
In the case of z, the number of quantization bits (Qb) = 24 bits, and 6 channels, a compression rate of 71% can be realized.

When the variable rate bit stream data predictively coded by the coding 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 includes pack start information of 4 bytes for user data (A packet and V packet) of 2034 bytes, and SCR (Syst of 6 bytes).
em Clock Reference: System time reference reference value information, 3-byte Mux rate information, and 1-byte stuffing, a total of 14-byte pack header is added (1 pack = 2048 bytes in total). . In this case, the SCR information, which is the time stamp,
In the first pack, "1" is set to be consecutive within the same title, so that the time of the A pack in the same title can be managed.

As shown in detail in FIG. 5, the compressed PCM A packet has a packet header of 9 to 22 bytes and a compressed P packet.
CM private header and 1 to 2015 bytes of audio data in the format shown in FIG. 3 (compressed P
CM). And DTS and PTS
Is set in the packet header of FIG. 5 (specifically, PTS is at the 10th to 14th bytes of the packet header and DTS is at the 15th to 19th bytes). The private header of the compressed PCM includes: 1-byte substream ID, 2-byte UPC / EAN-ISRC (Universal Pr
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 bytes of stuffing bytes. ing.

The forward access unit search pointer for searching the access unit one second later and the rear access unit search pointer for searching the access unit one second before are both 1 in the ADI.
Set in bytes. Specifically, the forward access unit search pointer is set at the 1st byte of the ADI and the backward access unit search pointer is set at the 8th byte. In this way, the ADI can store up to 2015 bytes of audio data because the compressed PCM is reduced to 4 bytes.

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. There is. 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", and a CR.
The substreams "BS0" and "BS1" are composed of C and extra information, and are composed of only PPCM blocks. The subpackets in the second and subsequent PPCM access units are also the directory, the substream "BS0", the CRC, and the substream "BS1".
And the CRC and the extra information, 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 adjusting function. That is, when the incoming data has a fixed rate (CBR), the sampling number per packet is 40, 80, 1 depending on the sampling frequency fs as described above.
It is set to any of 60, and therefore the data length per packet and the size of the subpacket may not match depending on the determined number of samplings. To match it with the size of the subpacket, for example, ,
Size adjustment is performed by adding 0, 0 ... Further, text data or the like can be used as the size adjusting data.

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. “0” (data indicating that the data in the subpacket is compressed data of VBR) when the data rate is VBR, and “1” (the data in the subpacket has a fixed rate) when the data rate is CBR Identifier indicating that) -sampling frequency fs and quantization bit number Qb-channel allocation information

Next, referring to FIG. 7, the decoding unit 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 ch
The leading sample data of one frame of "1" to "6" and the predictor selection flag are prediction circuits 24D1 and 24D, respectively.
2, 23D1 to 23D4 are applied to each channel “1” to
The bit number flag of “6” is applied to the unpacking circuit 22. The SCR, DTS, and prediction residual data string are applied to the input buffer 22 a, and PTS is applied to the output buffer 110. The identifiers indicating whether the data rate is VBR or CBR are the predictors 24D1 and 24D2,
Applied to 23D1, 23D2, 23D3, 23D4,
In these, the processing program of the input / output data according to the identifier is determined and processed. In the case of VBR, it is necessary to switch the processing program and load the input data every time, and thus it takes time to process, but in the case of CBR, since the fixed rate is used, the processing program does not need to be switched and the processing is performed. Get faster Further, the sampling frequency fs and the quantization bit number Qb are applied to the D / A converter 102. Here, the plurality of predictors (not shown) in the prediction circuits 24D1, 24D2, and 23D1 to 23D4 are the prediction circuits 1 on the encoding side, respectively.
The characteristics are the same as those of the plurality of predictors in 3D1, 13D2, 15D1 to 15D4, and those having the same characteristics are selected by the predictor selection flag.

The deformatting circuit 21 first separates the audio packet from the audio pack, and then separates the stream data (prediction residual data string) from the audio packet to form the bit streams BS0 and BS.
1 is taken out. Further, the SCR is taken out, and taken in and stored in 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, in the case of fs = 96 kHz (1/96 kHz)
Although it is seconds, it has a variable length as shown in detail in FIGS. 9 and 10A. Then, the stream data accumulated in the input buffer 22 a is read by the FIFO based on the DTS and applied to the unpacking circuit 22.

The unpacking circuit 22 has channels "1" to
The prediction residual data string of "6" is separated based on each bit number flag, and the prediction circuits 24D1, 24D2, and 23 are respectively separated.
Output to D1 to D4. Prediction circuits 24D1 and 24D
2, 23D1 to 23D4, the current prediction residual data of each channel “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 is added to calculate the current predicted value, and then the PC of each sample is based on the first sample data of one frame as a reference.
M data is calculated and stored in the output buffer 110. The PCM data stored in the output buffer 110 is P
It is read and output based on the TS, and thus the variable-length access unit shown in FIG. 10A is expanded and the constant-length presentation unit shown in FIG. 10B is output.

Further, based on the sampling frequency fs and the quantization bit number Qb in the PPCM sync information, the PC
The M data is converted into an analog signal by the D / A converter 102. At the same time, C in the PPCM sync information
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 it is subjected to decoding processing to be taken out 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 process is skipped. Also here
When the search reproduction is instructed through the operation unit 101, based on the front access unit search pointer (1 second ahead) and the rear access unit search pointer (1 second before) shown in FIG. Regenerate the access unit. For this search pointer, 1 second ahead,
Instead of 1 second ago, it may be 2 seconds ahead or 2 seconds ago.

When variable rate bit stream data predictively coded by the coding units 2'-1, 2'-2 shown in FIG. 2 are transmitted via a network, the coding side is as shown in FIG. Is packetized for transmission (step S41), a packet header is added (step S42), and this packet is then sent out on 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), this data is then stored in the memory and waits for decoding (step S53). . Then, when decoding is performed, as shown in FIG. 12B, the reformatting is performed (step S61), and then the input buffer 22
Input / output control of a is performed (step S62), and then 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 based on the flag to perform decoding (step S64), then input / output control of the output buffer 110 is performed (step S65), then the original multi-channel is restored (step S66), and then this is restored. It is output (step S67), and this is repeated thereafter.

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

As a third embodiment, as shown in FIG. 13, the stereo 2ch data (L, R) is downmixed by the formula (2) instead of 2ch "1", "2" to downmix the multichannel. The stereo 2ch (L, R) and the 4ch "3" to "6" may be generated and predictively encoded. Note that in the second and third embodiments, the front left (Lf) and the front right (Rf) are not transmitted to the decoding side, so the decoding side generates them by the equations (1) and (2).

Next, a fourth embodiment will be described with reference to FIGS. 14, 15 and 16. In the above embodiment,
Although it 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 predictively encoded, It is configured to select the predictive encoded data of the group having the highest compression rate. Further, in this embodiment, the coding within the one group is not performed by classifying and converting into 2ch related to the front group and 4ch related to other groups as in the case of each of the above-described embodiments. 1 is a configuration in which the encoding processing is put together into one.
4 is shown as a diagram corresponding to FIG. 1 described above. Also,
FIG. 15 shows a detailed block of the encoding unit.
In the case of this embodiment, n correlation circuits 1-1 to 1-n are provided on the side of the mix & matrix circuit 1 '.
These n correlation circuits 1-1 to 1-n are, for example, 6ch (L
The PCM data of f, C, Rf, Ls, Rs, and Lfe) 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 Further, the nth 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 with the highest compression ratio is based on the data amount of the minimum value of the prediction residual for each group. Are selected by the correlation selection signal generator 17b. At this time, the formatting circuit 19 adds the selection flag (correlation circuit selection flag, correlation coefficients a and b of the correlation circuit) and multiplexes them.

FIG. 16 shows a data area corresponding to the above-mentioned FIG. 6, and in this embodiment, the substream “B
S1 "is not used and only substream" BS0 "is used.

Also, on the decoding side shown in FIG. 17, n correlation circuits 4 are provided for the correlation 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, it is not necessary to provide the prediction circuits for n groups in the decoding device as shown in FIG. 17, and the prediction circuits for one group may be used. 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.
To set the original 6ch (Lf, C, Rf, Ls, Rs, L
fe) is restored, and the multi-channel is downmixed by the equation (2) to generate stereo 2ch data (L, R).

In the first embodiment, the signals "1" to "6" having one type of correlation are configured to be predictively coded, but the signals "1" to "6" are used. It is also possible to predictively code the group of and the group of the original signals (Lf, C, Rf, Ls, Rs, Lfe) and select the group having the higher compression rate.

[0035]

As described above, according to the present invention, the data packet is formatted into the data structure having the sub-packet containing the compressed data and the synchronization information part containing the sampling frequency and the number of quantization bits. When a multi-channel audio signal is encoded with a variable compression rate, the decoding efficiency on the reproducing side can be improved.

[Brief description of drawings]

FIG. 1 is a block diagram showing a first embodiment of a speech coding apparatus and speech decoding apparatus to which the present invention is applied .

FIG. 2 is a block diagram showing in detail a coding unit of FIG.

FIG. 3 is an explanatory diagram showing a bitstream encoded by the encoding unit 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 a format of an audio data area in FIG. 5 in detail.

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

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

FIG. 9 is an explanatory diagram showing a compressed data amount 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 showing a third embodiment of a speech coding apparatus and speech decoding apparatus to which the present invention is applied .

FIG. 14 is a block diagram showing a fourth embodiment of a speech coding apparatus and speech decoding apparatus to which the present invention is applied .

FIG. 15 is a block diagram showing a speech coding apparatus according to a fourth embodiment.

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

FIG. 17 is a block diagram showing a speech decoding apparatus according to a fourth embodiment.

[Explanation of symbols]

1'6ch mix & matrix circuit 13D1, 13D2, 15D1 to 15D4 Prediction circuit (buffer / selector 14D1, 14D2, 16D1 to 1)
A compression means is configured with 6D4. ) 14D1, 14D2, 16D1 to 16D4 buffers
Selector 17 Select 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

─────────────────────────────────────────────────── ─── Continuation of the front page (56) Reference JP-A-8-272393 (JP, A) JP-A-3-24834 (JP, A) JP-A-10-233058 (JP, A) JP-A-10- 320928 (JP, A) JP-A 64-44499 (JP, A) (58) Fields investigated (Int. Cl. ⁷ , DB name) G10L 19/04

Claims (4)

(57) [Claims] 1. A multichannel audio signal having a certain sampling frequency and a quantized bit number is obtained in response to an input audio signal for a channel as it is or for each channel correlated with each other, and a head sample value is obtained. Linear prediction values of the current signal are predicted from the past in the time domain by a plurality of linear prediction methods having different characteristics, and the prediction residual obtained from the predicted linear prediction value and the speech signal is minimized. A step of predictively encoding by selecting a linear prediction method; a linear prediction method for each channel selected in the step, a subpacket for storing prediction coded data including a prediction residual and a predetermined head sample value, and reproduction Side, a synchronization information part including a sampling frequency and a quantization bit number used when the original analog audio signal is restored. Formatting into a data structure according to the above, the data formatted in the data structure is recorded, and among the recorded data, the prediction coded data is a prediction value used for restoring the original audio signal. It is recorded as data for calculating be characterized Tei Rukoto
Recording medium . 2. The sub-packet and the synchronization information part according to claim 1, and a pack header including SCR information are further formatted and recorded as one pack, and the SCR information is recorded in the pack. A recording medium characterized by being used as time management information when reproducing. 3. A voice decoding device for decoding an original multi-channel voice signal from the data recorded on the recording medium according to claim 1, comprising means for separating the data structure into a subpacket and a synchronization information part. Decompressing means for decompressing the compressed data in the subpacket for each channel, means for converting the decompressed audio data for each channel into an original decompressed multichannel audio signal, and the converted multichannel Means for converting the audio data of 1. into an analog audio signal based on the sampling frequency and the number of quantization bits in the synchronization information part. 4. An audio decoding device for decoding an original multi-channel audio signal from the data recorded on the recording medium according to claim 2, wherein the first separating means separates the SCR information contained in the header. A buffer for holding the sub-packet and the synchronization information part based on the separated SCR information, and a second separating means for separating the sub-packet and the 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 part, and means for converting the decompressed audio data for each channel into the original multichannel decompressed audio signal. And converting the converted multi-channel audio data into an analog audio signal based on the sampling frequency and the number of quantization bits in the synchronization information section. And a means for converting into a speech signal.