JP4151020B2 - Audio signal transmission method and audio signal decoding apparatus - Google Patents

Abstract

<P>PROBLEM TO BE SOLVED: To enable 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: A speech signal transmitting method is a method of transmitting, in communication format of specified packets, data in data structure having an area of management information wherein a 1st identifier indicating whether multichannel data in an area (audio packet) of audio data are compressed, a 2nd identifier indicating whether down mixing of the multichannel data in the area (audio packet) of audio data into two stereophonic channels is permitted or inhibited, and a 3rd identifier indicating the table number of a down mixing coefficient for the down mixing and the effectiveness of the down mixing coefficient are arranged. Consequently, when data transmitted while the identifier indicating whether the 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&amp;NCIPI

Description

  The present invention relates to a multi-channel audio signal transmission method and a decoding apparatus thereof.

  As a method of compressing an audio signal with a variable length, the present inventor, in the previous application (Japanese Patent Application No. 9-289159), uses a plurality of predictors having different characteristics 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 speech signal and the plurality of linear prediction values, and a minimum prediction residual is selected. A predictive coding method is proposed.

  In the above method, a certain degree of compression effect can be obtained when the original digital audio signal has a sampling frequency = 96 kHz and the number of quantization bits = 20 bits. However, in recent DVD audio discs, the sampling frequency (twice this ( = 192 kHz), and the number of quantization bits tends to be 24 bits. In addition, the sampling frequency and the number of quantization bits in multichannel may be different for each channel.

  By the way, when transmitting multi-channel audio signals, the copyright holder may or may not want to compress depending on the audio source, and the user down-mixes the multi-channel to 2 stereo channels for playback. There are two ways: what you don't want to do and what you don't. Therefore, in the case of transmission in a total of four ways, such as two ways of selectively transmitting with compression or non-compression and two ways of selectively permitting and prohibiting down-mixing on the playback side, this is transmitted on the playback side. It is necessary to identify and selectively play back.

  Therefore, an object of the present invention is to provide an audio receiving method and an audio receiving apparatus that can be normally reproduced by the reproducing side even when the reproduction side downmix is selectively permitted or prohibited.

In order to achieve the above object, the present invention comprises the following means 1) to 3).
That is,
1) an audio packet in which compressed data or uncompressed data is selectively arranged in a multi-channel audio signal;
A first identifier indicating whether or not multi-channel data in the audio packet is compressed; and whether or not to allow the multi-channel data in the audio packet to be down-mixed to stereo two channels is indicated. An area of management information in which a second identifier, a downmix coefficient encoded for downmixing, and a third identifier indicating the validity of the downmix coefficient are arranged;
A voice signal transmission method comprising: packetizing data having a data structure having a packet and transmitting the data via a communication line .

2) an audio packet in which compressed data or uncompressed data is selectively arranged in a multi-channel audio signal;
A first identifier indicating whether or not multi-channel data in the audio packet is compressed; and whether or not to allow the multi-channel data in the audio packet to be down-mixed to stereo two channels is indicated. An area of management information in which a second identifier, a downmix coefficient encoded for downmixing, and a third identifier indicating the validity of the downmix coefficient are arranged;
A method of recording an audio signal, wherein the audio signal is encoded and recorded in a data structure.

3) A speech decoding apparatus for decoding data recorded by the recording method according to claim 2,
Means for separating the data into audio packets and management information;
When the second identifier in the management information permits down-mixing, the multi-channel data in the audio packet is expanded based on the first identifier, and the down-mix coefficient and the down-mix coefficient are valid. If the second identifier is prohibited from being downmixed by downmixing based on the third identifier indicating the sex and reproduced in stereo two channels, the multichannel data in the audio packet is Means for decompressing based on the identifier and playing only in multi-channel;
A speech signal decoding apparatus having:

  As described above, according to the present invention, for example, an identifier indicating whether or not multi-channel data is compressed, and whether or not to allow multi-channel data to be downmixed to two stereo channels are indicated. When receiving data transmitted by packetizing the identifier together, it can be decoded and reproduced normally.

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

Here, as the multi-channel method, for example, the following four methods are known.
(1) 4-channel system Like the Dolby Surround system, a total of 4 channels of 3 channels for the front L, C, and R + 1 channel for the rear S (2) 5 channels system Like no Dolby AC-3 system SW channel , Front L, C, R 3 channels + rear SL, SR 2 channels in total 5 channels (3) 6 channel system 6 channels (L, L, 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 forward L, LC, C, RC, R, and SW + 2 channels of backward SL and SR As an example of the transmission form, the case where the multi-channel is compressed and the reproduction side downmixing is prohibited is shown. The encoding-side 6-channel (ch) mix & matrix circuit 1 ′ includes, as an example of a multi-channel signal, a front left (Lf), a center (C), a front right (Rf), a surround left (Ls), and a surround right (Rs). ) And Lfe (Low Frequency Effect) 6ch PCM data is converted into correlation signals for 6ch “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)
The correlation equation by the 6-channel (ch) mix & matrix circuit 1 ′ and the encoding method of the encoding 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, and the selection means 7 'will be omitted in these drawings.

  The encoding unit 2 ′ having the first and second encoding units 2′-1, 2′-2 predictively encodes the 6ch “1” to “6” PCM data as shown in detail in FIG. Predictive encoded data is transmitted to the decoding side via the recording medium 5 and the communication medium 6 in a bit stream as shown in FIG. On the decoding side, the decoding unit 3 ′ having the first and second decoding units 3′-1 and 3′-2 performs prediction encoded data of 6ch “1” to “6” as shown in detail in FIG. Is then decoded into PCM data, and then only the original 6 channels (Lf, C, Rf, Ls, Rs, Lfe) are restored based on the equation (1-1) by the mix & matrix circuit 4 ′.

  FIG. 2 shows a case where the multi-channel is compressed and down-mixing on the reproduction side is permitted as a transmission form of the second example. The 6-channel mix & matrix circuit 1 'on the encoding side uses the original 6ch (Lf, C, Rf, Ls, Rs, Lfe) and the coefficient mij (i = 1, 2, j = 1, 2 to 6) as follows: As shown 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, the correlation signals “1” and “2” for the two channels of the first group and the correlation signals “3” for the four channels of the second group as shown below by the equation (2) and the following equation (1-2): 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 formula (1-1) (1-2)
The first and second encoding units 2′-1 and 2′-2 predictively encode the PCM data of the first group channels “1” and “2” and the second group channels “3” to “6”, respectively. Predictive encoded 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 respectively perform predictive coding of the first group channels “1” and “2” and the second group channels “3” to “6”. The data is decoded into PCM data, and then the original 6ch (Lf, C, Rf, Ls, Rs, Lfe) is restored based on the equations (1-2) and (2) by the mix & matrix circuit 4 ′. Stereo 2ch data (L, R) is generated by adding and subtracting the first group channels “1” and “2”, respectively.

  FIG. 3 shows a case where the multi-channel is transmitted without compression and the reproduction side down-mixing is prohibited as a third example of transmission form. In this case, since it is uncompressed, the encoding side transmits the original 6ch (Lf, C, Rf, Ls, Rs, Lfe) PCM data as it is without generating a correlation signal (however, formatting) On the decoding side, after deformatting, only the original 6ch (Lf, C, Rf, Ls, Rs, Lfe) is restored.

  FIG. 4 shows a transmission example of the fourth example in which multi-channel transmission is performed without compression and down-mixing on the reproduction side is permitted. Also in this case, since the data is not compressed, the encoding side transmits the original 6ch (Lf, C, Rf, Ls, Rs, Lfe) PCM data without generating a correlation signal for increasing the compression rate. (But format). On the decoding side, after deformatting, the original 6ch (Lf, C, Rf, Ls, Rs, Lfe) is restored, and stereo 2ch data (L, R) is generated (downmix) according to equation (2) To do.

  FIG. 5 shows a modification in the case where the multi-channel is compressed in FIG. 1 and down-mixing on the reproduction side is prohibited. In this case, the encoding side converts the correlation signals for 6ch (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 6ch (Lf, C, Rf, Ls, Rs, Lfe) is restored by Expression (1-2).

“1” = Lf−C
“2” = Rf−C
“3” to “6” are the same as in formula (1-1) (1-3)
Thus, when the reproduction side down-mixing is prohibited, the down-mix coefficient of the equation (2) is not added to the encoding correspondingly, and the stereo 2ch data (L, It is forbidden to generate (downmix) R).

  FIG. 6 shows a modification in the case of compressing the multi-channel in FIG. 2 and allowing down-mixing on the reproduction side. In this case, on the encoding side, stereo 2ch data (L, R) is generated (downmixed) according to the equation (2), and then the following two channels of the first group “ 1 ”,“ 2 ”and the correlation signals“ 3 ”to“ 6 ”for the four channels of the second group, and the first and second encoding units 2′-1, 2′-2 Is predictively encoded. Then, on the decoding side, the original 6ch (Lf, C, Rf, Ls, Rs, Lfe) is restored by the 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 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 channel “1” to “6” is stored in one frame buffer 10 for each frame. The sample data of each channel “1” to “6” of one frame is applied to the prediction circuits 13D1, 13D2, and 15D1 to 15D4, respectively, and the head sample data of each frame of each channel “1” to “6” Is applied to the formatting circuit 19. Each of the prediction circuits 13D1, 13D2, 15D1 to 15D4 outputs a current signal from a past signal in the time domain to a PCM data of each channel “1” to “6” by a plurality of predictors (not shown) having different characteristics. A plurality of linear prediction values 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 subsequent buffer / selectors 14D1, 14D2, 16D1 to 16D4 temporarily store the prediction residuals calculated by the prediction circuits 13D1, 13D2, and 15D1 to 15D4, respectively, and select signals / DTS (decoding time stamp). The minimum value of the prediction residual is selected for each subframe designated by the generator 17.

  The selection signal / DTS generator 17 applies the bit number flag of the prediction residual to the packing circuit 18 and the formatting circuit 19, and the predictor selection flag indicating the predictor with the smallest prediction residual and the phase relationship The number a and the DTS indicating the time for the decoding side to extract 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 6ch selected by the buffers / selectors 14D1, 14D2, 16D1 to 16D4 with the designated number of bits based on the bit number flag designated by the selection signal / DTS generator 17. . The PTS generator 17c generates a PTS (Presentation Time Stamp) indicating the time for the decoding side to extract the PCM data from the output buffer 110 (FIG. 14) and outputs the PTS to the formatting circuit 19. The formatting circuit 19 is also applied with an encoding mode indicating compression / non-compression and an identifier indicating downmix permission / prohibition.

  The subsequent formatting circuit 19 formats the user data as shown in FIGS. The user data (subpacket) shown in FIG. 8 includes variable rate bitstream (substream) BS0 including 2ch “1” and “2” predictive encoded data regarding the forward group, and 4ch “3” to “3” regarding other groups. It is composed of a variable rate bit stream (substream) BS1 including predictive encoded data of “6” and a bitstream header (restart header) provided before the substreams BS0 and BS1.

Also, one frame of substream BS0, BS1 is a frame header,
・ First sample data of one frame of each channel “1” to “6”,
A predictor selection flag for each subframe of each channel “1” to “6”;
A bit number flag for each subframe of each channel “1” to “6”;
-Predictive residual data string (number of variable bits) of each ch "1" to "6",
The coefficient a of ch “6” is
Multiplexed. According to such predictive coding, when the original signal is, for example, sampling frequency = 96 kHz, quantization bit number = 24 bits, and 6 channels, a compression rate of 71% can be realized.

  When the variable rate bit stream data predictively encoded by the encoding units 2′-1, 2′-2 shown in FIG. 7 is recorded on a DVD audio disk as an example of a recording medium, the audio ( A) Packed in a pack. This pack consists of 2034 bytes of user data (A packet, V packet), 4 bytes of pack start information, 6 bytes of SCR (System Clock Reference) information, and 3 bytes of Mux rate ( rate) information and a 1-byte stuffing total 14-byte pack header are added (1 pack = total 2048 bytes). In this case, the time of the A pack in the same title can be managed by setting the SCR information as a time stamp as “1” in the first pack and continuing in the same title.

As shown in FIG. 10, the compressed PCM A packet is composed of a 19 or 14 byte packet header, a compressed PCM private header, and 1 to 2011 byte audio data (compressed PCM) in the format shown in FIG. ing. The DTS and PTS are set in the packet header of FIG. 5 (specifically, the PTS is in the 10th to 14th bytes of the packet header and the DTS is in the 15th to 19th bytes). The compressed PCM private header is
A 1-byte substream ID,
-2-byte UPC / EAN-ISRC (Universal Product Code / European Article Number-International Standard Recording Code) number and UPC / EAN-ISRC data;
-1 byte private header length,
A 2-byte first access unit pointer;
8 bytes of audio data information (ADI)
・ With stuffing byte of 0-7 bytes,
It is made up of.

  In addition, a front access unit search pointer for searching for an access unit after one second and a rear access unit search pointer for searching for an access unit one second before are both set in one byte in the ADI. 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 is composed of subpackets 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. Has been. The 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. The substreams “0” and “1” are PPCM. It consists only of blocks. Sub-packets in the second and subsequent PPCM access units are composed of sub-stream “0”, CRC, sub-stream “1”, CRC and extra information except for the directory, and sub-streams “0”, “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 (an identifier indicating that the data in the subpacket is compressed data)
-Sampling frequency fs and number of quantization bits Qb
Channel assignment information The 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 AOTT-AOB-ATR (audio only title / audio object set attribute), and this AOTT-AOB-ATR (b127 to b0) is an 8-bit (b127 to b120) audio code in order from the MSB side. Mode,
An 8-bit (b119 to b112) reserved area;
A quantization bit number Q1 of a channel group “1” of 4 bits (b111 to b108);
A quantization bit number Q2 of a channel group “2” of 4 bits (b107 to b104);
A sampling frequency fs1 of a channel group “1” of 4 bits (b103 to b100);
A sampling frequency fs2 of a 4-bit (b99 to b96) channel group “2”;
3 bit (b95 to b93) multi-channel structure type;
-Channel assignment of 5 bits (b92 to b88);
-It is composed 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 encoding 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) Quantization bit number Q2 of channel group 2 (b107 to b104)
When the number of quantization bits Q1 of channel group 1 is “0000b”, “0000b”
“0000b” or “0001b” when the number of quantization bits Q1 of channel group 1 is “0001b”
“0000b”, “0001b” or “0010b” when the number of quantization bits Q1 of channel group 1 is “0010b”
However, 0000b: 16 bits
0001b: 20 bits
0010b: 24 bits
Other: Reserved (4) Channel group 1 sampling frequency fs1 (b103 to b100) 0000b: 48 kHz
0001b: 96 kHz
0010b: 192 kHz
1000b: 44.1 kHz
1001b: 88.2 kHz
1010b: 176.4 kHz
Other: Reserved (5) Sampling frequency fs2 (b99 to b96) of channel group 2
“0000b” when the sampling frequency fs1 of channel group 1 is “0000b”
When the sampling frequency fs1 of the channel group 1 is “0001b”, “0000b” or “0001b”
When the sampling frequency fs1 of the channel group 1 is “0010b”, “0000b”, “0001b” or “0010b”
“1000b” when the sampling frequency fs1 of channel group 1 is “1000b”
When the sampling frequency fs1 of the channel group 1 is “1001b”, “1000b” or “1001b”
When the sampling frequency fs1 of the channel group 1 is “1010b”, “1000b”, “1001b”, or “1010b”
(6) Type of multi-channel structure (b95 to b93)
000b: Type 1
Other: 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 between the previous and current PG (R / A),
1-bit (b30) STC discontinuity flag (STC-F);
・ The number of attributes (ATRN) of 3 bits (b29 to b27),
3 bit (b26 to b24) channel group (ChGr) “2” bit shift data;
A 2-bit (b23, b22) reserved area;
1-bit (b21) downmix mode (D-M),
・ Effectiveness of 1-bit (b20) downmix coefficient (shown *),
A 4-bit (b19 to b16) downmix coefficient table number (DM-COEFTN);
Each is composed of RTI flags F15 to F0 of 1 bit and 16 bits (b15 to b0) in total.

  When the downmix mode (DM) of the bit (b21) is “1”, “downmix prohibition” is indicated, and when it is “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 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 first sample data of one frame of each channel “1” to “6” and the predictor selection flag are respectively applied to the prediction circuits 24D1, 24D2, 23D1 to 23D4, and the number of bits of each channel “1” to “6”. The flag is applied to the unpacking circuit 22. The SCR, DTS, and prediction residual data string are applied to the input buffer 22a, and 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, and 15D1 to 15D4, respectively. The same characteristic is selected by the predictor selection flag.

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

  Further, the PCM data is converted into an analog signal by the D / A converter 102 based on the sampling frequency fs and the quantization bit number Qb in the PPCM sync information. Here, when search reproduction is instructed via the operation unit 101, the control unit 100 causes the front access unit search pointer (one second ahead) and the rear access unit search pointer (one second previous) shown in FIG. Play the access unit based on This search pointer may be one second ahead and two seconds ahead instead of one second ahead and one second ahead.

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

  As shown in FIG. 19A, the decoding side removes the header (step S51), then restores the data (step S52), then stores this data in the memory and waits for decoding (step S53). When decoding is performed, as shown in FIG. 19B, deformatting is performed (step S61), input / output control of the input buffer 22a is performed (step S62), and then unpacking is performed (step S61). S63). At this time, if there is a search reproduction instruction, the search pointer is decoded. Next, the predictor is selected and decoded based on the flag (step S64), then the input / output control of the output buffer 110 is performed (step S65), and then the original multi-channel is restored (step S66). This is repeated (step S67).

  Next, a second embodiment will be described with reference to FIGS. In the above embodiment, a group of correlated signals “1” to “6” is configured to be predictively encoded. In the fourth embodiment, a plurality of groups of correlated signals are generated. Thus, the prediction encoding is performed, and the prediction encoded data of the group having the highest compression rate is selected. Therefore, the encoding unit shown in FIG. 20 includes first to nth correlation circuits 1-1 to 1-n. The n correlation circuits 1-1 to 1-n are, for example, 6ch (Lf, C , Rf, Ls, Rs, Lfe) is 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 converts as follows.

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

  Further, on the decoding side shown in FIG. 21, n correlation circuits 4-1 to 4-n (or coefficients a and b can be changed to 1 with respect to the correlation circuits 1-1 to 1-n on the encoding side. Two correlation circuits 4) are provided. When the n groups of prediction circuits shown in FIG. 20 have the same configuration, the decoding device does not need to have n groups of prediction circuits as shown in FIG. 21, and only one group of prediction circuits 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 set and the original 6ch (Lf, C, Rf, Ls, Rs, Lfe) is restored, and the multi-channel is downmixed according to Equation (2) to generate stereo 2ch data (L, R).

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

  According to the present invention, in addition to the invention described in the claims, the following invention is provided.

Formatting means for selectively placing compressed or uncompressed data in a multi-channel audio signal in an audio packet;
Depending on whether the multi-channel data in the audio packet is compressed or whether the multi-channel data in the audio packet is allowed to be down-mixed to stereo 2 channels or not, the code is down-mixed in advance. Means for selecting whether or not to encode, or whether to encode downmix coefficients,
A speech encoding apparatus having the same.

It is explanatory drawing which shows the 1st example of the transmission form of the multichannel to which this invention is applied. It is explanatory drawing which shows the 2nd example of the transmission form of the multichannel to which this invention is applied. It is explanatory drawing which shows the 3rd example of the transmission form of the multichannel to which this invention is applied. It is explanatory drawing which shows the 4th example of the transmission form of the multichannel to which this invention is applied. It is explanatory drawing which shows the modification of FIG. It is explanatory drawing which shows the modification of FIG. It is a block diagram which shows the encoding part of FIG. 1 in detail. It is explanatory drawing which shows the bit stream encoded by the encoding part of FIG. 1, FIG. It is explanatory drawing which shows the format of the pack of DVD. It is explanatory drawing which shows the format of the audio pack of DVD. It is explanatory drawing which shows the format of the audio data area of FIG. 10 in detail. It is explanatory drawing which shows AOTT-AOB-ATR (audio only title / audio object set attribute) of DVD audio. It is explanatory drawing which shows ATS-PG-CNT (audio title set program content) of DVD audio. It is a block diagram which shows the decoding part of FIG. 1 in detail. 15 is a timing chart showing write / read timings of the input buffer of FIG. It is explanatory drawing which shows the compressed data amount for every access unit. It is explanatory drawing which shows an access unit and a presentation unit. It is a flowchart which shows the audio | voice transmission method. It is a flowchart which shows the audio | voice transmission method. It is a block diagram which shows the audio | voice coding apparatus of 2nd Embodiment. It is a block diagram which shows the audio | voice decoding apparatus of 2nd Embodiment.

Explanation of symbols

1 '6ch mix & matrix circuit 13D1, 13D2, 15D1-15D4 Prediction circuit (compressing means is constituted with buffer / selectors 14D1, 14D2, 16D1-16D4)
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 (expanding means)
100 Control unit (reproducing means)
102 D / A converter 110 Output buffer

Claims (3)

Audio packets in which multi-channel audio signals are compressed or uncompressed data are selectively placed;
A first identifier indicating whether or not multi-channel data in the audio packet is compressed; and whether or not to allow the multi-channel data in the audio packet to be down-mixed to stereo two channels is indicated. An area of management information in which a second identifier, a downmix coefficient encoded for downmixing, and a third identifier indicating the validity of the downmix coefficient are arranged;
A voice signal transmission method comprising: packetizing data having a data structure having a packet and transmitting the data via a communication line . Audio packets in which multi-channel audio signals are compressed or uncompressed data are selectively placed;
A first identifier indicating whether or not multi-channel data in the audio packet is compressed; and whether or not to allow the multi-channel data in the audio packet to be down-mixed to stereo two channels is indicated. An area of management information in which a second identifier, a downmix coefficient encoded for downmixing, and a third identifier indicating the validity of the downmix coefficient are arranged;
A method of recording an audio signal, wherein the audio signal is encoded and recorded in a data structure. 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;
When the second identifier in the management information permits down-mixing, the multi-channel data in the audio packet is expanded based on the first identifier, and the down-mix coefficient and the down-mix coefficient are valid. If the second identifier is prohibited from being downmixed by downmixing based on the third identifier indicating the sex and reproduced in stereo two channels, the multichannel data in the audio packet is Means for decompressing based on the identifier and playing only in multi-channel;
A speech signal decoding apparatus having:

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