JP2004226991A - Speech signal transmitting method and speech signal decoding device - Google Patents

Abstract

<P>PROBLEM TO BE SOLVED: To enables a reproduction side to perform reproduction normally even when multiple channels are selectively transmitted on a compression or non-compression basis or down mixing on the reception side is selectively permitted or inhibited. <P>SOLUTION: ATSI includes a 1st identifier showing whether multichannel data in an audio packet are compressed or not and a 2nd identifier indicating whether down mixing of the multichannel data into two stereophonic channels is permitted or inhibited. Thus, even when data transmitted while the identifier showing whether or not multichannel data are compressed and the identifier indicating whether down mixing of the multichannel data into two stereophonic channels is permitted or inhibited are both packetized are received, the data can be decoded and reproduced normally. <P>COPYRIGHT: (C)2004,JPO&NCIPI

Description

  The present invention relates to a method for transmitting a multi-channel audio signal and a decoding device therefor.

  As a method of compressing an audio signal with a variable length, the present inventor has proposed in the earlier application (Japanese Patent Application No. 9-289159) that a plurality of predictors having different characteristics are used in the time domain for one channel of the original digital audio signal. A plurality of linear prediction values of the current signal are calculated from the past signal, a prediction residual for each predictor is calculated from the original digital audio signal and the plurality of linear prediction values, and a minimum value of the prediction residual is selected. A prediction coding method is proposed.

  In the above method, a certain compression effect can be obtained when the original digital audio signal has a sampling frequency = 96 kHz and the number of quantization bits = approximately 20 bits. = 192 kHz) 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.

  By the way, when transmitting a multi-channel audio signal, the copyright holder may or may not desire compression depending on the audio source, and the user may down-mix the multi-channel to two stereo channels for playback. There are two types of things that you do not want to do and those that do 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 is performed on the reproduction side. It needs to be identified and selectively played.

  Therefore, an object of the present invention is to provide an audio receiving method and an audio receiving apparatus that allow the reproducing side to normally reproduce even if the downmix on the reproducing side is selectively permitted or prohibited.

The present invention comprises the following means 1) to 3) to achieve the above object.
That is,
1) an audio data area (audio packet) in which multi-channel audio signal compressed data or uncompressed data is selectively arranged;
A first identifier indicating whether or not the multi-channel data in the audio data area (audio packet) is compressed; and down-mixing the multi-channel data in the audio data area (audio packet) into two stereo channels. An area of management information in which a second identifier indicating whether to permit or prohibit the operation and a third identifier indicating the structure of the multi-channel are arranged;
An audio signal transmission method characterized by transmitting data having a data structure having a communication format of a predetermined packet.

2) a region (audio packet) of audio data in which compressed data 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 data area (audio packet) is compressed; and down-mixing the multi-channel data in the audio data area (audio packet) into two stereo channels. An area of management information in which a second identifier indicating whether to permit or prohibit the operation and a third identifier indicating the structure of the multi-channel are arranged;
A sound signal recording method characterized in that the sound signal is encoded and recorded in a data structure having the following.

3) An audio decoding device for decoding data recorded by the recording method according to claim 2,
Means for separating the data into audio packets and management information, wherein the multi-channel data in the audio packets is based on the first identifier if a second identifier in the management information allows downmixing. The multi-channel data in the audio packet is reproduced based on the first identifier when the reproduction is performed on the multi-channel and the stereo two-channel with or without the expansion and the down-mix identifier prohibits the down-mix. Means for selectively decompressing or not decompressing and identifying a channel by the third identifier and reproducing only with multi-channels;
An audio signal decoding device having:

  As described above, according to the present invention, for example, an identifier indicating whether or not multi-channel data is compressed and whether to permit or prohibit down-mixing of multi-channel data to two stereo channels are indicated. In the case of receiving data transmitted with the identifier packetized together, the data can be normally decoded and reproduced.

  Hereinafter, embodiments of the present invention will be described 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, for example, the following four systems are known as the multi-channel system.
(1) Four-channel system Like the Dolby surround system, three channels of front L, C, and R + one channel of rear S are total of four channels. (2) Five-channel system As with no Dolby AC-3 system SW channel. , Front L, C, R 3 channels + rear SL, SR 2 channels, total 5 channels (3) 6 channel system 6 channels (L, L, C) such as DTS (Digital Theater System) system and Dolby AC-3 system 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 front L, LC, C, RC, R, and SW + 2 channels of rear SL and SR are shown in FIG. In this example, a case is shown in which the multi-channel is compressed and the downmix on the playback side is prohibited. The 6-channel (ch) mix & matrix circuit 1 ′ on the encoding side includes front left (Lf), center (C), front right (Rf), surround left (Ls), and surround right (Rs) as examples of multi-channel signals. ) And Lfe (Low Frequency Effect) 6-channel PCM data are converted into correlation signals for 6 channels “1” to “6” by the following equation (1-1) and output to the encoding unit 2 ′.

"1" = Lf + Rf-C
"2" = Lf-Rf-C
“3” = C− (Ls + Rs) / 2
"4" = Ls + Rs
“5” = Ls−Rs
“6” = Lfe−a × C
However, 0 ≦ a ≦ 1 (1-1)
Such a 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 means 7 '. The same applies to FIG. 2, FIG. 3, FIG. 4, FIG. 5, and FIG. 6, which will be described below. Therefore, in these figures, the selection means 7 'is omitted.

  The encoding unit 2 ′ including the first and second encoding units 2′-1 and 2′-2 predictively encodes the PCM data of 6ch “1” to “6” as shown in detail in FIG. The prediction coded 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, as shown in detail in FIG. 14, the prediction coded data of 6ch "1" to "6" is decoded by the decoding unit 3 'having the first and second decoding units 3'-1 and 3'-2. Is decoded into PCM data, and only the original 6 channels (Lf, C, Rf, Ls, Rs, Lfe) are restored by the mix & matrix circuit 4 ′ based on the equation (1-1).

  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 uses the original 6 channels (Lf, C, Rf, Ls, Rs, Lfe) and the coefficients mij (i = 1, 2, j = 1, 2 to 6) as follows: As described in (2), stereo 2ch data (L, R) is generated (downmixed).

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 the equation (2) and the following equation (1-2), the correlation signals “1” and “2” for two channels of the first group and the correlation signal “3” for four channels of the second group are as follows. To "6" and output to the first encoding unit 2'-1 and the second encoding unit 2'-2, respectively.

"1" = L + R
"2" = LR
"3" to "6" are the same as in the formula (1-1) ... (1-2)
The first and second encoding units 2′-1 and 2′-2 predictively encode PCM data of the first group channels “1” and “2” and the second group channels “3” to “6”, respectively. The prediction coded data of each channel is transmitted to the decoding side via the recording medium 5 and 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) is restored by the mix & matrix circuit 4 ′ based on the equations (1-2) and (2). Stereo 2-ch data (L, R) is generated by adding and subtracting the first 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 the data is uncompressed, the encoding side transmits the original 6-channel (Lf, C, Rf, Ls, Rs, Lfe) PCM data as it is without generating a correlation signal (however, the formatting is performed). Then, on the decoding side, after reformatting, only the original 6 channels (Lf, C, Rf, Ls, Rs, Lfe) are restored.

  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. Also in this case, since the data is uncompressed, the encoding side transmits the original 6-channel (Lf, C, Rf, Ls, Rs, Lfe) PCM data without generating a correlation signal for increasing the compression rate. Yes (but format). On the decoding side, after performing the reformatting, the original 6 ch (Lf, C, Rf, Ls, Rs, Lfe) is restored, and the stereo 2ch data (L, R) is generated by equation (2) (downmix). I do.

  FIG. 5 shows a modification in which the multi-channel is compressed and the downmix on the reproduction side is prohibited in FIG. In this case, on the encoding side, the signal is converted into a correlation signal for 6 channels (1) to (6) by the following equation (1-3), and the encoding unit 2 'performs predictive encoding. Then, on the decoding side, only the original 6 channels (Lf, C, Rf, Ls, Rs, Lfe) are restored by equation (1-2).

"1" = Lf-C
"2" = Rf-C
"3" to "6" are the same as in the formula (1-1) ... (1-3)
In the case where the downmix on the reproduction side is prohibited in this manner, the downmix coefficient of the equation (2) is not added to the encoding corresponding to this, and the stereo 2ch data (L, R) is forbidden.

  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 encoding units 2'-1 and 2'-2 convert the respective group channels. Is predictively coded. Then, on the decoding side, the original 6 channels (Lf, C, Rf, Ls, Rs, Lfe) are restored according to equations (1-4) and (2), and the stereo 2ch data (L, R) is output as it is.

"1" = L
"2" = R
"3" to "6" are the same as in the formula (1-1) ... (1-4)
The encoding units 2′-1 and 2′-2 will be described in detail with reference to FIG. The PCM data of each of the channels “1” to “6” is stored in one frame buffer 10 for each frame. Then, the sample data of each channel “1” to “6” of one frame is applied to the prediction circuits 13D1, 13D2, 15D1 to 15D4, respectively, and the first sample data of each frame of each channel “1” to “6” Is applied to the formatting circuit 19. The prediction circuits 13D1, 13D2, and 15D1 to 15D4 respectively use a plurality of predictors (not shown) having different characteristics to convert the current signal from the past signal in the time domain to the PCM data of each channel “1” to “6”. Are calculated, and then a prediction residual for each predictor is calculated from the original PCM data and the plurality of linear prediction values. The following buffer / selectors 14D1, 14D2, 16D1 to 16D4 temporarily store the prediction residuals calculated by the prediction circuits 13D1, 13D2, 15D1 to 15D4, respectively, and provide a 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 furthermore, a predictor selection flag indicating the predictor having the minimum prediction residual and a phase relationship. The number a and the DTS indicating the time at which the decoding side extracts the stream data from the input buffer 22a (FIG. 14) are applied to the formatting circuit 19. 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. . The PTS generator 17c generates a PTS (presentation time stamp) indicating the time at which the decoding side takes out the PCM data from the output buffer 110 (FIG. 14), and outputs the PTS to the formatting circuit 19. 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 formats the user data as shown in FIGS. The user data (sub-packet) shown in FIG. 8 includes a variable-rate bit stream (sub-stream) BS0 including 2ch “1” and “2” prediction coded data for the front group, and 4ch “3” to 4ch for the other groups. It is composed of a variable-rate bit stream (substream) BS1 containing the prediction encoded data of “6” and a bitstream header (restart header) provided before the substreams BS0 and BS1.

Also, one frame of the substreams BS0 and BS1 has a frame header,
-First sample data of one frame of each channel "1" to "6";
A predictor selection flag for each subframe of each of the channels “1” to “6”;
A bit number flag for each subframe of each channel “1” to “6”;
A prediction residual data string (variable number of bits) for each channel “1” to “6”;
-The coefficient a of ch "6" is
It is multiplexed. According to such predictive coding, when the original signal has, for example, a sampling frequency of 96 kHz, a quantization bit number of 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, the audio data shown in FIG. A) Packed in a pack. This pack has 4 bytes of pack start information, 20 bytes of SCR (System Clock Reference: System Clock Reference) information, and 3 bytes of Mux rate (20 bytes) of user data (A packet, V packet). rate) information and 1-byte stuffing, and a pack header of 14 bytes in total is 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 to “1” in the first pack and making it continuous within the same title.

As shown in detail in FIG. 10, the A packet of the compressed PCM is composed of a packet header of 19 or 14 bytes, a private header of the compressed PCM, and 1 to 2011 bytes of audio data (compressed PCM) in the format shown in FIG. ing. Then, the DTS and the PTS are set in the packet header of FIG. 5 (specifically, the PTS is set at the 10th to 14th bytes and the DTS is set at the 15th to 19th bytes). The compressed PCM private header is
A 1-byte substream ID,
A 2-byte UPC / EAN-ISRC (Universal Product Code / European Article Number-International Standard Recording Code) number and UPC / EAN-ISRC data;
A 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 composed of

  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 are set in the ADI in one byte. Specifically, the forward access unit search pointer is set to the seventh byte of the ADI, and the backward access unit search pointer is set to the eighth byte.

  The audio data area in the compressed PCM (also referred to as PPCM) audio packet shown in FIG. 10 is composed of a subpacket and a plurality of PPCM access units as shown in FIG. 11, and the PPCM access unit is composed of PPCM sink information and subpackets. Have been. A subpacket in the first PPCM access unit is composed of a directory, a substream “0”, a CRC, a substream “1”, a CRC and extra information, and the substreams “0” and “1” are PPCMs. It is composed of only blocks. The sub-packets in the second and subsequent PPCM access units are composed of a sub-stream “0”, a CRC, a sub-stream “1”, a CRC and extra information except for a directory, and sub-streams “0” and “1”. Is composed of 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" in the case of VBR (identifier indicating that the data in the subpacket is compressed data)
.Sampling frequency fs and quantization bit number Qb
The channel assignment information formatting circuit 19 also uses an ATSI (Audio Title Set Information) including management information as shown in FIGS. 12 and 13 to manage the audio packs shown in FIGS. Format. FIG. 12 shows an AOTT-AOB-ATR (audio only title audio object set attribute). The AOTT-AOB-ATR (b127 to b0) is an 8-bit (b127 to b120) audio code in order from the MSB side. Activation mode,
An 8-bit (b119 to b112) reserved area;
A 4-bit (b111 to b108) channel group “1” quantization bit number Q1;
A 4-bit (b107 to b104) quantization bit number Q2 of the channel group “2”;
A sampling frequency fs1 of a 4-bit (b103 to b100) channel group "1";
A sampling frequency fs2 of a 4-bit (b99 to b96) channel group “2”;
A 3-bit (b95 to b93) multi-channel structure type;
Channel assignment of 5 bits (b92 to b88);
-Consists of a reserved area of 8 bits x 11 (b87 to b0).

The above data is shown in detail below.
(1) Audio encoding mode (b127 to b120)
00000000b: Linear PCM mode 00000001b: Compressed PCM mode Others: reserved for other coding modes (2) Quantization bit number Q1 of channel group 1 (b111 to b108)
0000b: 16 bits 0001b: 20 bits 0010b: 24 bits Others: reserved (3) Number of quantization bits Q2 of channel group 2 (b107 to b104)
"0000b" when the quantization bit number Q1 of the channel group 1 is "0000b"
"0000b" or "0001b" when the quantization bit number Q1 of the channel group 1 is "0001b"
When the quantization bit number Q1 of the channel group 1 is “0010b”, “0000b”, “0001b” or “0010b”
However, 0000b: 16 bits
0001b: 20 bits
0010b: 24 bits
Others: reserved (4) Sampling frequency fs1 of channel group 1 (b103 to b100) 0000b: 48 kHz
0001b: 96 kHz
0010b: 192 kHz
1000b: 44.1 kHz
1001b: 88.2 kHz
1010b: 176.4 kHz
Others: reserved (5) Sampling frequency fs2 of channel group 2 (b99 to b96)
"0000b" when the sampling frequency fs1 of the channel group 1 is "0000b"
"0000b" or "0001b" when the sampling frequency fs1 of the channel group 1 is "0001b"
"0000b", "0001b" or "0010b" when the sampling frequency fs1 of the channel group 1 is "0010b"
"1000b" when the sampling frequency fs1 of the channel group 1 is "1000b"
"1000b" or "1001b" when the sampling frequency fs1 of the channel group 1 is "1001b"
When the sampling frequency fs1 of the channel group 1 is “1010b”, “1000b”, “1001b”, or “1010b”
(6) Multi-channel structure type (b95 to b93)
000b: Type 1
Others: reserved (7) Channel allocation (b92 to b88)
Channel assignment information of groups “1” and “2” from 1 channel (monaural) to 6 channels FIG. 13 shows ATS-PG-CNT (audio title set program content), which is 1 bit in order from the top (B31) the relationship (R / A) between the previous and current PGs,
1-bit (b30) STC discontinuity flag (STC-F);
The number of attributes (ATRN) of 3 bits (b29 to b27);
3 bits (b26 to b24) of a channel group (ChGr) “2” bit shift data;
A 2-bit (b23, b22) reserved area;
1-bit (b21) downmix mode (DM);
・ Effectiveness (illustration *) of 1-bit (b20) downmix coefficient,
A 4-bit (b19 to b16) downmix coefficient table number (DM-COEFTN);
-Each bit is composed of RTI flags F15 to F0 each having 1 bit and a total of 16 bits (b15 to b0).

  When the downmix mode (DM) of the bit (b21) is “1”, “downmix prohibition” is indicated, and when “0”, “downmix permission” is indicated.

  Next, the decoding unit 3 '(3'-1, 3'-2) will be described with reference to FIG. The decoding section 3 '(3'-1, 3'-2) and the mix & matrix circuit 4' can be realized not only by hardware but also by a computer program. The variable rate bit stream data BS0 and BS1 in the above format are separated by the deformatting circuit 21. Then, the head sample data of one frame of each channel “1” to “6” and the predictor selection flag are applied to the prediction circuits 24D1, 24D2, 23D1 to 23D4, respectively, and the bit numbers of each channel “1” to “6” The flag 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 plurality of predictors (not shown) in the prediction circuits 24D1, 24D2, 23D1 to 23D4 have the same characteristics as the plurality of predictors in the encoding-side prediction circuits 13D1, 13D2, 15D1 to 15D4, respectively. , And those having the same characteristics are selected by the predictor selection flag.

  The stream data (predicted residual data string) separated by the deformatting circuit 21 is fetched and accumulated 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 (1/96 kHz) seconds when fs = 96 kHz, for example, but has a variable length as shown in detail in FIGS. 16 and 17A. 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 sequence of each of the channels “1” to “6” based on the bit number flag and outputs the separated data to the prediction circuits 24D1, 24D2, and 23D1 to 23D4. The prediction circuits 24D1, 24D2, and 23D1 to 23D4 respectively use the current prediction residual data of each of the channels “1” to “6” from the unpacking circuit 22 and a predictor selection flag among a plurality of internal predictors. The previous predicted value predicted by each selected one is added to calculate the current predicted value, and then the PCM data of each sample is calculated based on the first sample data of one frame and stored in the output buffer 110. Is done. The PCM data stored in the output buffer 110 is read out and output based on the PTS. Therefore, the variable-length access unit shown in FIG. 17A is expanded and the fixed-length access unit shown in FIG. The presentation unit is output.

  The PCM data is converted into an analog signal by the D / A converter 102 based on the sampling frequency fs and the number of quantization bits Qb in the PPCM sync information. 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 ago) 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 encoded by the encoding unit 2 ′ (2′-1, 2′-2) is transmitted via a network, the encoding side uses a transmission-rate bit stream as shown in FIG. (Step S41), a packet header is added (step S42), and the packet is sent out to 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 and waiting for decoding (step S53). When decoding is performed, as shown in FIG. 19B, deformatting is performed (step S61), input / output control of the input buffer 22a is performed (step S62), and then unpacking is performed (step S62). 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 and decoding is performed (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 coding and select predictive coded data of a group having the highest compression ratio. For this reason, the encoding unit shown in FIG. 20 is provided with first to n-th correlation circuits 1-1 to 1-n, and the n correlation circuits 1-1 to 1-n have, for example, 6 channels (Lf, Cf). , Rf, Ls, Rs, and Lfe) are converted into n types of 6-channel signals “1” to “6” having different correlations.

For example, the first correlation circuit 1-1 converts as follows,
(1) = Lf
(2) = C− (Ls + Rs) / 2
(3) = Rf-Lf
(4) = Ls−a × Lfe
(5) = Rs−b × Rf
(6) = Lfe
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 a group having the highest compression rate is selected for correlation based on the data amount of the minimum value of the prediction residual for each group. Selected by the 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).

  In addition, on the decoding side shown in FIG. 21, n correlation circuits 4-1 to 4-n (or coefficients a and b in which the number of changeable coefficients a and b are One correlation circuit 4) is provided. When the prediction circuits of the n groups shown in FIG. 20 have the same configuration, the decoding device does not need to provide the prediction circuits of the n groups as shown in FIG. 21, but may use the prediction circuits of one group. Then, one of the correlation circuits 4-1 to 4-n is selected based on the selection flag transmitted from the encoding device, or the coefficients a and b are set and the original 6 ch (Lf, C, Rf, Ls, Rs, Lfe) are 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 correlated signals “1” to “6” is configured to be predictively coded. A group of original signals (Lf, C, Rf, Ls, Rs, Lfe) may be predictively coded and a group having a higher compression rate 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;
The code is downmixed beforehand depending on whether the multi-channel data in the audio packet is compressed, or whether to allow or prohibit the down-mixing of the multi-channel data in the audio packet into two stereo channels. Means to select whether to encode, or whether to encode the downmix coefficient,
Speech encoding device having the same.

FIG. 3 is an explanatory diagram showing a first example of a multi-channel transmission mode to which the present invention is applied. FIG. 7 is an explanatory diagram showing a second example of a multi-channel transmission mode to which the present invention is applied. FIG. 9 is an explanatory diagram showing a third example of a multi-channel transmission mode to which the present invention is applied. FIG. 11 is an explanatory diagram showing a fourth example of a multi-channel transmission mode to which the present invention is applied. FIG. 4 is an explanatory diagram showing a modification of FIG. 1. FIG. 6 is an explanatory diagram showing a modification of FIG. 2. FIG. 2 is a block diagram illustrating an encoding unit of FIG. 1 in detail. FIG. 8 is an explanatory diagram illustrating a bit stream encoded by the encoding unit in FIGS. 1 and 7. FIG. 4 is an explanatory diagram showing a format of a DVD pack. FIG. 3 is an explanatory diagram showing a format of a DVD audio pack. FIG. 11 is an explanatory diagram showing the format of the audio data area in FIG. 10 in detail. FIG. 3 is an explanatory diagram showing AOTT-AOB-ATR (audio only title, audio object set, attribute) of DVD audio. FIG. 4 is an explanatory diagram showing ATS-PG-CNT (audio title set program content) of DVD audio. FIG. 2 is a block diagram illustrating a decoding unit of FIG. 1 in detail. 15 is a timing chart showing write / read timings of the input buffer of FIG. FIG. 4 is an explanatory diagram showing a compressed data amount for each access unit. FIG. 3 is an explanatory diagram showing an access unit and a presentation unit. 5 is a flowchart illustrating a voice transmission method. 5 is a flowchart illustrating a voice transmission method. It is a block diagram showing a speech encoding device of a second embodiment. It is a block diagram showing a speech decoding device of a second embodiment.

Explanation of reference numerals

1 '6ch Mix & Matrix Circuit 13D1, 13D2, 15D1 to 15D4 Prediction circuit (compresses with buffer / selector 14D1, 14D2, 16D1 to 16D4)
14D1, 14D2, 16D1 to 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 Prediction circuit (expansion means)
100 control unit (reproduction means)
102 D / A converter 110 Output buffer

Claims (3)

An audio data area (audio packet) in which compressed data 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 data area (audio packet) is compressed; and down-mixing the multi-channel data in the audio data area (audio packet) into two stereo channels. An area of management information in which a second identifier indicating whether to permit or prohibit the operation and a third identifier indicating the structure of the multi-channel are arranged;
An audio signal transmission method characterized by transmitting data having a data structure having a communication format of a predetermined packet. An audio data area (audio packet) in which compressed data 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 data area (audio packet) is compressed; and down-mixing the multi-channel data in the audio data area (audio packet) into two stereo channels. An area of management information in which a second identifier indicating whether to permit or prohibit the operation and a third identifier indicating the structure of the multi-channel are arranged;
A sound signal recording method characterized in that the sound signal is encoded and recorded in a data structure having the following. An audio decoding device for decoding data recorded by the recording method according to claim 2,
Means for separating the data into audio packets and management information, wherein the multi-channel data in the audio packets is based on the first identifier if a second identifier in the management information allows downmixing. The multi-channel data in the audio packet is reproduced based on the first identifier when the reproduction is performed on the multi-channel and the stereo two-channel with or without the expansion and the down-mix identifier prohibits the down-mix. Means for selectively decompressing or not decompressing and identifying a channel by the third identifier and reproducing only with multi-channels;
An audio signal decoding device having:

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