JP3196776B1 - Audio encoding device, audio transmission method, and computer program recording medium - Google Patents

Abstract

The present invention enables a reproduction side to perform normal reproduction even when a multi-channel is selectively transmitted in a compressed or uncompressed state, or selectively permits or prohibits a downmix on the reproduction side. A first identifier indicating whether or not the multi-channel data in the audio packet is compressed; and a second identifier indicating whether to allow or prohibit the down-mixing of the multi-channel data into two stereo channels. .

Description

DETAILED DESCRIPTION OF THE INVENTION

[0001]

The present invention relates to a multi-channel audio encoding device, an audio transmission method, and a recording medium on which a computer program for decoding audio data is recorded.

[0002]

2. Description of the Related Art As a method of compressing an audio signal with a variable length, the present inventor has filed a prior application (Japanese Patent Application No. 9-289159).
Calculates a plurality of linear prediction values of a current signal from a past signal in the time domain by using a plurality of predictors having different characteristics with respect to the one-channel original digital audio signal. A prediction encoding method for calculating a prediction residual for each predictor from a prediction value and selecting a minimum value of the prediction residual has been 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]

When transmitting a multi-channel audio signal, the copyright holder may or may not desire compression depending on the audio source. There are two types, one that does not want to downmix and play back to stereo two channels and one that does not. Therefore, if the transmission is performed in two ways, that is, selectively transmitted with compression or non-compression and the downmix on the reproduction side is selectively permitted and prohibited, a total of four transmissions are performed on the reproduction side. It must be identified and selectively played.

[0005] Therefore, the present invention provides an audio coding system that allows the reproducing side to reproduce normally even if the multi-channel is selectively transmitted with compression or non-compression, or the downmix on the reproducing side is selectively permitted or prohibited. It is an object of the present invention to provide an apparatus, an audio transmission method, and a recording medium on which a computer program for decoding audio data is recorded.

[0006]

The present invention, in order to achieve the above object, comprises the following 1) to 3). That is, 1) an audio packet in which data of which a multi-channel audio signal is compressed or uncompressed data is selectively arranged, and a first identifier indicating whether or not the multi-channel data in the audio packet is compressed. When,
Means for formatting into a data structure having management information arranged with a second identifier indicating whether to allow or prohibit downmixing of the multi-channel data in the audio packet into two stereo channels. a speech coding apparatus comprising, when said second identifier indicating the permission of the down-mix a stereo 2 multichannel data in the audio packet
Encoding downmix coefficients for downmixing to a channel , wherein the second identifier inhibits downmixing.
If so, do not encode the downmix coefficients.
Speech coding apparatus characterized by Uni. 2) an audio packet in which data in which a multi-channel audio signal is compressed or uncompressed data is selectively arranged, a first identifier indicating whether or not the multi-channel data in the audio packet is compressed;
Communication by packetizing the data in the data structure having a multi-channel management information data and the second identifier indicating whether or prohibiting allowed to down-mix stereo 2 channels are arranged in said audio packet A voice transmission method characterized by transmitting via a line. 3) an audio packet in which compressed or uncompressed data of a multi-channel audio signal is selectively arranged; a first identifier indicating whether or not the multi-channel data in the audio packet is compressed;
Said recording medium and a multi-channel data to the stereo 2-channel and a second identifier indicating whether or prohibited are allowed to down-mix is placed management information in the audio packet, the data structure having is recorded A computer program recording medium in which a computer program for decoding data recorded in is recorded,
The computer program includes the steps of separating the data into audio packets and management information, and extracting the first identifier and the second identifier from the management information, the second identifier extracted a downmix when indicating the permission decodes not selectively elongating or stretching based multi-channel data in the audio packet to a first identifier the extracted,
Removed at least one of the multi-channel and stereo 2 channels, when said second identifier indicates that the prohibited <br/> stop the down-mix, the first identifier multichannel data in the audio packet And selectively decompressing and decoding without decompression based on the multi-channel method based on a multi-channel.

[0007]

Embodiments of the present invention will be described below with reference to the drawings. FIG. 1 to FIG. 4 are explanatory diagrams showing the processing of a speech coding apparatus for realizing a multi-channel transmission mode to which the present invention is applied.

Here, as the multi-channel system, for example, the following four systems are known. (1) Four-channel system As in the Dolby surround system, a total of four channels including three channels of front L, C, and R + one channel of rear S (2) Five-channel system S in the Dolby AC-3 system
Like 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) like 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

FIG. 1 shows, as a transmission form of the first example, a case in which multi-channels are compressed and downmixing on the reproduction side is prohibited. 6 channels (ch) on the encoding side
The mix & matrix circuit 1 ′ includes a front left (Lf), a center (C), a front right (Rf), a surround left (Ls), a surround right (Rs), and an Lfe (Low Frequency Ef) as an example of a multi-channel signal.
fect) of 6 ch by the following equation (1-1).
The signals are converted into correlation signals for ch "1" to "6",
Output to “1” = Lf + Rf−C “2” = Lf−Rf−C “3” = C− (Ls + Rs) / 2 “4” = Ls + Rs “5” = Ls−Rs “6” = Lfe−a × C 0 ≦ a ≦ 1 (1-1) The correlation formula by the 6-channel (ch) mix & matrix circuit 1 ′ and the coding system of the coding unit 2 ′ are selected by the selection unit 7 ′. 2, 3, and 4 described below.
Since the same applies to FIGS. 5 and 6, the selecting means 7 'is omitted in these figures.

First and second encoding units 2'-1, 2'-2
The encoding unit 2 ′ having
The PCM data of ch “1” to “6” is predictively coded, and the predicted coded data is transmitted to the decoding side via the recording medium 5 or the communication medium 6 as a bit stream as shown in FIG. On the decoding side, the decoding unit 3 ′ having the first and second decoding units 3′-1 and 3′-2 causes the 6-ch
The predictive coded data of "1" to "6" is decoded into PCM data, and then the formula (1) is mixed by the mix & matrix circuit 4 '.
-1) based on the original 6 ch (Lf, C, Rf, Ls, R
s, Lfe) only.

FIG. 2 shows, as a transmission form of the second example, a case in which multi-channels are compressed and downmixing on the reproduction side is permitted. The 6-channel mixing and matrix circuit 1 ′ on the encoding side converts the original 6-channel (Lf, C, Rf, Ls,
Rs, Lfe) and coefficient mij (i = 1, 2, j = 1, 2 to
According to 6), stereo 2ch data (L, R) is generated (downmixed) 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)

Then, according to equations (2) and (1-2), correlation signals "1" and "2" for two channels of the first group and correlation signals for four channels of the second group are obtained as follows. Are converted to “3” to “6”, and are respectively encoded by the first encoding unit 2 ′.
−1, output to the second encoding unit 2′-2. “1” = L + R “2” = LR “3” to “6” are the same as the formula (1-1) ... (1-2)

The first and second encoding units 2'-1 and 2'-2 predict the PCM data of the first group channels "1" and "2" and the second group channels "3" to "6", respectively. The encoded data is transmitted to the decoding side via the recording medium 5 or the communication medium 6. On the decoding side, the first and second decoding units 3'-1 and 3'-2 predictively encode the first group channels "1" and "2" and the second group channels "3" to "6", respectively. The data is decoded into PCM data, and the original 6 ch (Lf, C,
Rf, Ls, Rs, Lfe) and restore the first
Stereo 2-ch data (L, R) is generated by adding and subtracting the group channels “1” and “2”.

FIG. 3 shows, as a transmission form of the third example, a case where multi-channel transmission is performed without compression and downmixing on the reproduction side is prohibited. In this case,
Since it is uncompressed, the encoding side does not generate a correlation signal and the original 6 ch (Lf, C, Rf, Ls, Rs, Lf
e) The PCM data of (e) is transmitted as it is (however, it is formatted) and the decoding side performs deformatting, and then the original 6 ch (Lf, C, Rf, Ls, Rs, Lfe)
Restore only.

FIG. 4 shows, as a transmission form of the fourth example, a case where multi-channel transmission is performed without compression and downmixing on the reproduction side is permitted. Again, in this case,
Since it is uncompressed, the encoding side does not generate a correlation signal for increasing the compression ratio without generating the original 6ch (Lf, C, R
f, Ls, Rs, Lfe) PCM data is transmitted as it is (formatted). On the decoding side, after reformatting, the original 6 ch (Lf, C, Rf,
Ls, Rs, and Lfe) are restored, and stereo 2ch data (L, R) is generated (downmixed) by equation (2).

FIG. 5 shows a modification of FIG. 1 in which the multi-channel is compressed and the downmix on the reproduction side is prohibited. In this case, the encoding side converts into correlation signals for 6 channels (1) to (6) according to the following equation (1-3), and the encoding unit 2 'predictively encodes the signals. And
On the decoding side, the original 6 ch (Lf, C,
Rf, Ls, Rs, Lfe) alone. “1” = Lf−C “2” = Rf−C “3” to “6” are the same as Expression (1-1)... (1-3) When prohibiting the downmix on the reproduction side, Correspondingly, the downmix coefficient of equation (2) is not added to the encoding, and the encoding side performs stereo 2 by equation (2).
Generation (downmixing) of the ch data (L, R) is prohibited.

FIG. 6 shows a modification in which the multi-channel is compressed and the downmix on the reproduction side is permitted in FIG. In this case, the encoding side generates (down-mixes) stereo 2ch data (L, R) according to equation (2), and then, according to equation (1-4), the first group of two channels “ 1 "," 2 "and correlation signals" 3 "to" 6 "for four channels of the second group, and the first and second encoders 2'-1 and 2'-2 convert the respective group channels. Is predictively encoded. Then, on the decoding side, the original 6 ch (Lf, C, R) is obtained by the equations (1-4) and (2).
f, Ls, Rs, Lfe) are restored, and the stereo 2ch data (L, R) is output as it is. “1” = L “2” = R “3” to “6” are the same as the formula (1-1) ... (1-4)

Referring to FIG. 7, 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
The first sample data of each frame of “1” to “6” is applied to the formatting circuit 19. Prediction circuit 13D
1, 13D2, 15D1 to 15D4 are each channel
For the PCM data of “1” to “6”, a plurality of linear prediction values of a current signal are calculated from a past signal in a time domain by a plurality of predictors (not shown) having different characteristics, and then the original PCM data Then, a prediction residual for each predictor is calculated from the plurality of linear prediction values. Following buffer / selector 14D
1, 14D2, 16D1 to 16D4 temporarily store the prediction residuals calculated by the prediction circuits 13D1, 13D2, 15D1 to 15D4, respectively, and select the selection signal / DTS.
(Decoding Time Stamp) The minimum value of the prediction residual is selected for each subframe specified by the 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. , The correlation coefficient a, and the DTS indicating the time at which the decoding side extracts the stream data from the input buffer 22a (FIG. 14).
To be applied. The packing circuit 18 packs the prediction residuals for the 6 channels selected by the buffer / selectors 14D1, 14D2, 16D1 to 16D4 with the specified number of bits based on the bit number flag specified by the selection signal / DTS generator 17. . Also, the PTS generator 17c
The decoding circuit generates a PTS (presentation time stamp) indicating a time at which the PCM data is taken out from the output buffer 110 (FIG. 14), and formats the PTS.
9 is output. An encoding mode indicating compression / non-compression and an identifier indicating downmix permission / prohibition are also applied to the formatting circuit 19.

The following formatting circuit 19 is shown in FIGS.
Formatted into user data as shown in FIG. The user data (sub-packet) shown in FIG. 8 is a variable-rate bit stream (sub-stream) BS including 2ch “1” and “2” prediction coded data for the front group.
0, a variable-rate bit stream (sub-stream) BS1 including 4ch “3” to “6” prediction coded data relating to other groups, and sub-streams BS0 and BS1
Is composed of a bit stream header (restart header) provided before.

Also, 1 of the substreams BS0 and BS1
The frame portion includes: 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 sub-frame of ch “1” to “6”; a prediction residual data string (variable bit number) of each ch “1” to “6”; and a coefficient a of ch “6” Are multiplexed. According to such predictive coding, when the original signal has, for example, a sampling frequency of 96 kHz, the number of quantization bits = 24 bits, and 6 channels, a compression ratio of 71% can be realized.

When recording the variable rate bit stream data predictively coded by the coding units 2'-1 and 2'-2 shown in FIG. 7 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 to “1” so as to be continuous within the same title.

The A packet of the compressed PCM has a 19 or 14 byte packet header, as shown in detail in FIG.
It is composed of a compressed PCM private header and 1 to 2011 bytes of audio data (compressed PCM) in the format shown in FIG. And DTS
The PTS is included in the packet header of FIG. 5 (specifically, the PTS is 15 to 1 in the 10th to 14th bytes of the packet header).
DTS is set at the ninth byte. 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 bytes of first access unit pointer, 8 bytes of audio data information (ADI), and 0 to 7 bytes of stuffing bytes. ing.

Also, a forward access unit search pointer for searching for an access unit one second later and a backward access unit search pointer for searching for an access unit one second earlier in the ADI are both set to one byte. Is done. Specifically, the forward access unit search pointer is set in the seventh byte of the ADI, and the backward access unit search pointer is set in the eighth byte.

The audio data area in the audio packet of the compressed PCM (also referred to as PPCM) shown in FIG. 10 includes a subpacket and a plurality of PPs as shown in FIG.
It is composed of a CM access unit, and the PPCM access unit is composed of PPCM sink information and subpackets. The sub-packets in the first PPCM access unit are: directory, sub-stream “0”, CRC, sub-stream “1”, CRC
The substreams “0” and “1” are composed of only PPCM blocks. The sub-packets in the second and subsequent PPCM access units are, except for the directory, sub-stream “0”, CRC, sub-stream “1”, CRC
The substreams “0” and “1” are configured by a restart header and a PPCM block.

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. -Data rate: "0" for VBR (identifier indicating that data in a subpacket is compressed data)-Sampling frequency fs and number of quantization bits Qb-Channel allocation information

The formatting circuit 19 also manages the audio packs shown in FIGS.
The ATSI (Audio Title Set Information) including the management information as shown in FIG. 13 is formatted. FIG. 12 shows AOTT-AOB-ATR
(Audio only title, audio object set attribute), and the AOTT-AO
The B-ATR (b127 to b0) includes, from the MSB side, an audio coding mode of 8 bits (b127 to b120); a reserved area of 8 bits (b119 to b112); and a 4-bit (b111 to b108). The number of quantization bits Q1 of the channel group “1” of the above; the number of quantization bits Q2 of the channel group “2” of 4 bits (b107 to b104); and the group of the bits 1 of 4 bit (b103 to b100) A 4-bit (b99 to b96) sampling frequency fs2 of the channel group "2"; a 3-bit (b95 to b93) multi-channel structure type; and a 5-bit (b92 to b88). And a reservation area of 8 bits × 11 (b87 to b0).

The above data is shown below in detail. (1) Audio encoding mode (b127 to b120) 00000000b: Linear PCM mode 00000001b: Compressed PCM mode Others: reserved for other encoding modes

(2) Number of quantized bits Q1 of channel group 1 (b111 to b108) 0000b: 16 bits 0001b: 20 bits 0010b: 24 bits Other: reserved (3) Number of quantized bits Q2 (b1) of channel group 2
07-b104) The quantization bit number Q1 of the channel group 1 is “000”
0b "is" 0000b ". The number of quantization bits Q1 of the channel group 1 is" 000 ".
1b "is" 0000b "or" 0001b ". The quantization bit number Q1 of the channel group 1 is" 001 ".
In the case of “0b”, “0000b”, “0001b” or “0010b” where 0000b: 16 bits 0001b: 20 bits 0010b: 24 bits Other: reserved

(4) Sampling frequency fs1 of channel group 1 (b103 to b100) 0000b: 48
kHz 0001b: 96kHz 0010b: 192kHz 1000b: 44.1kHz 1001b: 88.2kHz 1010b: 176.4kHz Other: reserved

(5) Sampling frequency fs2 of channel group 2 (b99 to b96)-"0000b" when sampling frequency fs1 of channel group 1 is "0000b"-Sampling frequency fs1 of channel group 1 of "0001b" In this case, "0000b" or "000
1b ”•“ 0000b ”,“ 0001 ”when the sampling frequency fs1 of the channel group 1 is“ 0010b ”
"b" or "0010b"-"1000b" when the sampling frequency fs1 of the channel group 1 is "1000b"-"1000b" or "100" when the sampling frequency fs1 of the channel group 1 is "1001b"
1b ”-“ 1000b ”,“ 1001 ”when the sampling frequency fs1 of the channel group 1 is“ 1010b ”
b "or" 1010b "

(6) Multi-channel structure type (b9
000b: Type 1 Other: Reserved (7) Channel allocation (b92 to b88) Channel allocation information of groups "1" and "2" from 1 channel (monaural) to 6 channels

FIG. 13 shows ATS-PG-CNT (audio title set program content),
This is, in order from the beginning: 1 bit (b31), the relationship between the previous and current PG (R
/ A) and 1-bit (b30) STC discontinuity flag (STC
-F), the number of attributes of three bits (b29 to b27) (A
TRN) and a 3-bit (b26 to b24) channel group (C
hGr) "2" bit shift data; 2 bit (b23, b22) reserved area; 1 bit (b21) downmix mode (D-
M), 1-bit (b20) downmix coefficient validity (illustrated *), 4-bit (b19-b16) downmix coefficient table number (DM-COEFTN), 1 bit each, It consists of RTI flags F15 to F0 of a total of 16 bits (b15 to b0). Then, the downmix mode (DM) of the bit (b21)
Is "1", "downmix prohibition", and "0" indicates "downmix permission".

Next, referring to FIG. 14, the decoding unit 3 '(3'
-1, 3'-2) will be described. The decoding section 3 '(3'-1, 3'-2) and the mix & matrix circuit 4' can be realized by a computer program in addition to hardware. The variable rate bit stream data BS0 and BS1 in the above format are separated by the deformatting circuit 21. The head sample data of one frame of each of the channels “1” to “6” and the predictor selection flag are respectively stored in the prediction circuits 24D1 and 24D.
D2, 23D1 to 23D4, 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 encoding mode indicating compression / non-compression and an identifier indicating downmix permission / prohibition are applied to the control unit 100, and the sampling frequency fs and the number of quantization bits Qb are applied to the D / A converter 102. . Here, the prediction circuits 24D1, 24D2, 23D1
A plurality of predictors (not shown) in the 23D4 are prediction circuits 13D1, 13D2, 15D1 to 15D on the encoding side, respectively.
4 have the same characteristics as the plurality of predictors, and those having the same characteristics are selected by the predictor selection flag.

The stream data (prediction residual data string) separated by the reformatting circuit 21 is fetched and stored in the input buffer 22a for each access unit by SCR as shown in FIG. Here, the data amount of one access unit is, for example, fs = 96 kHz.
In the case of z, it is (1/96 kHz) seconds, but FIG.
6, variable length as shown in detail in FIG. 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. 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. 17A is expanded, and a fixed-length presentation unit shown in FIG. 17B 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. Here, when a search reproduction is instructed via the operation unit 101, the control unit 100 causes the forward access unit search pointer (one second ahead) and the backward access unit search pointer (one second ahead) shown in FIG. Play the access unit based on the. The search pointer may be one second ahead, two seconds ahead, two seconds ahead instead of one second ahead.

When variable-rate bit stream data predictively coded by the coding unit 2 '(2'-1, 2'-2) is transmitted via a network, the coding side uses the data as shown in FIG. Into a packet for transmission (step S41), and then add a packet header (step S41).
42), and then sends this packet out onto the network (step S43).

On the decoding side, as shown in FIG. 19A, 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. 19B, 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.

Next, a second embodiment will be described with reference to FIGS. In the above embodiment, 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. Then, it is configured to perform predictive encoding and select predictive encoded data of a group having the highest compression ratio. Therefore, the encoding unit shown in FIG. 20 includes first to n-th correlation circuits 1-1 to 1-n,
The n correlation circuits 1-1 to 1-n have, for example, 6 channels (L
f, C, Rf, Ls, Rs, and Lfe) PCM data 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 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 and multiplexes the selection flag (correlation circuit selection flag, correlation coefficients a and b of the correlation circuit).

On the decoding side shown in FIG. 21, n correlating circuits 4 are provided to correlating circuits 1-1 to 1-n on the encoding side.
−1 to 4-n (or one correlation circuit 4 whose coefficients a and b can be changed) are provided. When the prediction circuits of the n groups shown in FIG. 20 have the same configuration,
As shown in FIG. 1, there is no need to provide prediction circuits for n groups, and prediction circuits for one group are sufficient. 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 set and the original 6 ch (Lf, C, Rf, Ls, Rs, Lfe) is restored, and multi-channels are downmixed 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. According to the present invention, the following inventions are provided in addition to the inventions described in the claims. Formatting means for selectively arranging compressed data or uncompressed data of a multi-channel audio signal in an audio packet; and determining whether multi-channel data in the audio packet is compressed, or Means for selecting whether to perform down-mixing and encoding in advance or whether to encode down-mix coefficients according to whether to allow or prohibit down-mixing of multi-channel data into two stereo channels. An audio encoding device comprising:

[0045]

As described above, according to the present invention, for example, when a multi-channel is selectively transmitted with compression or non-compression, an identifier indicating whether or not multi-channel data is compressed, Since the channel data is formatted into a data structure having an identifier indicating whether to permit or prohibit the downmixing into two stereo channels, even if the downmix on the playback side is selectively permitted or prohibited, The reproducing side can reproduce normally.

[Brief description of the drawings]

FIG. 1 is an explanatory diagram showing a first example of a multi-channel transmission mode to which the present invention is applied.

FIG. 2 is an explanatory diagram showing a second example of a multi-channel transmission mode to which the present invention is applied.

FIG. 3 is an explanatory diagram showing a third example of a multi-channel transmission mode to which the present invention is applied.

FIG. 4 is an explanatory diagram showing a fourth example of a multi-channel transmission mode to which the present invention is applied.

FIG. 5 is an explanatory diagram showing a modification of FIG. 1;

FIG. 6 is an explanatory diagram showing a modification of FIG. 2;

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

FIG. 8 is an explanatory diagram showing a bit stream encoded by the encoding unit shown in FIGS. 1 and 7;

FIG. 9 is an explanatory diagram showing a format of a DVD pack.

FIG. 10 is an explanatory diagram showing a format of a DVD audio pack.

11 is an explanatory diagram showing the format of the audio data area in FIG. 10 in detail.

FIG. 12 DVD audio AOTT-AOB-AT
It is explanatory drawing which shows R (audio only title, audio object set attribute).

FIG. 13 ATS-PG-CNT of DVD audio
FIG. 4 is an explanatory diagram showing (audio title set program content).

FIG. 14 is a block diagram illustrating a decoding unit of FIG. 1 in detail.

FIG. 15 is a timing chart showing write / read timings of the input buffer of FIG. 14;

FIG. 16 is an explanatory diagram showing the amount of compressed data for each access unit.

FIG. 17 is an explanatory diagram showing an access unit and a presentation unit.

FIG. 18 is a flowchart showing a voice transmission method.

FIG. 19 is a flowchart illustrating an audio transmission method.

FIG. 20 is a block diagram illustrating a speech encoding device according to a second embodiment.

FIG. 21 is a block diagram illustrating a speech decoding device according to a second 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 17c PTS generator 19 Formatting circuit 21 Deformatting circuit (separating means) 22 Unpacking circuit 22a Input buffer 24D1, 24D2, 23D1 to 23D4 Predicting circuit (expanding means) 100 Control unit ( Reproduction means) 102 D / A converter 110 Output buffer

────────────────────────────────────────────────── ─── Continuation of the front page (56) References JP-A-9-259539 (JP, A) JP-A-10-21673 (JP, A) (58) Fields investigated (Int. Cl. ⁷ , DB name) G11B 20/10 H04S 3/00 H03M 7/30 H04B 14/04

Claims (3)

(57) [Claims] An audio packet in which compressed or uncompressed data of a multi-channel audio signal is selectively arranged, and a first indicating whether or not the multi-channel data in the audio packet is compressed. Formatting a data structure having an identifier and management information in which a second identifier indicating whether to allow or prohibit downmixing of the multi-channel data in the audio packet into two stereo channels is arranged. a speech coding apparatus comprising means for, the when the second identifier indicating the permission of the down-mix is downmix coefficients for down-mix the multi-channel data to stereo 2 channels in the audio packet Encoding the said
When the identifier of 2 indicates that downmix is prohibited
Is a device for not encoding the downmix coefficient . 2. An audio packet in which compressed data or uncompressed data of a multi-channel audio signal is selectively arranged, and a first indicating whether or not the multi-channel data in the audio packet is compressed. A packet of data having a data structure having an identifier and a management information in which a second identifier indicating whether to allow or prohibit downmixing of the multi-channel data in the audio packet into two stereo channels is disposed. A voice transmission method characterized in that the voice transmission is performed through a communication line. 3. An audio packet in which data in which a multi-channel audio signal is compressed or data in which it is not compressed is selectively arranged, and a first indicating whether the multi-channel data in the audio packet is compressed or not. Management information in which an identifier and a second identifier indicating whether to allow or prohibit downmixing the multi-channel data into two stereo channels in the audio packet are arranged;
A computer program recording medium on which a computer program for decoding data recorded on a recording medium on which a data structure is recorded is recorded, the computer program comprising: separating the data into audio packets and management information; , extracting the first identifier and the second identifier from the management information, the second identifier extracted permits the downmix
To the case shown, the decoded without or extension selectively extending the basis of multi-channel data in the audio packet to a first identifier the extracted at least one of the multi-channel and stereo 2 channels extraction, this said second identifier prohibit downmix
To the case shown TMG, and removing the multi-channel data in the audio packet only multichannel decoded without or extension selectively extending the basis of the first identifier comprises a computer program storage medium .