JP4983847B2 - Audio signal transmission device, audio signal reception device, and audio signal transmission system - Google Patents

Description

The present invention relates to an audio signal transmission device, an audio signal reception device, and an audio signal transmission system for multi-channel audio signals.

As a method of compressing an audio signal with a variable length, the present inventor has proposed an earlier application (Japanese Patent Application No. 9-2
89159), for a single channel original digital audio signal, a plurality of linear prediction values of the current signal are calculated from past signals in the time domain by using a plurality of predictors having different characteristics. A prediction encoding method is proposed in which a prediction residual for each predictor is calculated from a plurality of linear prediction values and a minimum value of the prediction residual is selected.

In the above method, the original digital audio signal has a sampling frequency of 96 kHz,
A certain degree of compression effect can be obtained when the number of quantization bits is about 20 bits, but in recent DVD audio discs, the sampling frequency (=
192 kHz) is used, 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.
Accordingly, an object of the present invention is to provide an audio signal transmission device, an audio signal reception device, and an audio signal transmission system that can be normally reproduced by the reproduction side even if the reproduction side downmix is selectively permitted or prohibited. To do.

In order to achieve the above object, the present invention comprises the following means 1) to 3).
That is,
1) Correlation between two channels of downmix processing channels, which are channels obtained by correlating two or more multi-channel audio signals obtained by downmixing, and a plurality of channels For each channel, the first sample value is obtained in units of frames for a predetermined time in response to the input audio signal, and a plurality of prediction units having different characteristics are used to obtain the current sample from the past in the time domain. A prediction unit that obtains a linear prediction value such that a prediction residual obtained from the predicted linear prediction value and the speech signal is minimized by further predicting a linear prediction value of each of the signals. Select by subframe and predictively encode,
The data structure includes a pack header including SCR information and user data including a compressed PCM access unit, and a plurality of the compressed PCM access units are provided in the frame. A first bitstream including the two-channel downmix processing channels arranged in the compressed PCM access unit, and prediction encoded data including prediction unit selection information indicating a prediction unit of each channel and a prediction residual; The predicted residual is packed and stored in a sub-packet composed of a second bit stream including other channels with the number of bits based on the designated bit information, and the data in the sub-packet is compressed with a variable bit rate. VBR identifier indicating that it is compressed data and the original analog on the playback side A synchronization information part including a sampling frequency and the number of quantization bits used for restoration to a voice signal is provided, and the head sample value is stored in the subpacket in the compressed PCM access unit. An audio signal transmission apparatus for transmitting data encoded by an audio encoding method,
An audio signal transmission apparatus characterized in that the encoded encoded data including the head sample value, prediction unit selection information indicating the selected prediction unit, and prediction residual and the SCR information are packetized and transmitted.
2) Correlation between two channels of downmix processing channels, which are channels obtained by correlating each other between two channels obtained by downmixing three or more multichannel audio signals, and a plurality of channels For each channel, the first sample value is obtained in units of frames for a predetermined time in response to the input audio signal, and a plurality of prediction units having different characteristics are used to obtain the current sample from the past in the time domain. A prediction unit that obtains a linear prediction value such that a prediction residual obtained from the predicted linear prediction value and the speech signal is minimized by further predicting a linear prediction value of each of the signals. Select by subframe and predictively encode,
The data structure includes a pack header including SCR information and user data including a compressed PCM access unit, and a plurality of the compressed PCM access units are provided in the frame. A first bitstream including the two-channel downmix processing channels arranged in the compressed PCM access unit, and prediction encoded data including prediction unit selection information indicating a prediction unit of each channel and a prediction residual; The predicted residual is packed and stored in a sub-packet composed of a second bit stream including other channels with the number of bits based on the designated bit information, and the data in the sub-packet is compressed with a variable bit rate. VBR identifier indicating that it is compressed data and the original analog on the playback side A synchronization information part including a sampling frequency and the number of quantization bits used for restoration to a voice signal is provided, and the head sample value is stored in the subpacket in the compressed PCM access unit. Receiving means for receiving a packet in which data encoded by a voice encoding method is packetized and transmitted;
Restoring means for restoring original data based on the received packet;
Time management means for managing the compressed PCM access unit of the restored data by SCR information;
Extracting a VBR identifier from the synchronization information part, taking a leading sample value from the subpacket based on the extracted identifier and taking out a prediction residual and a prediction part selection information indicating a prediction part from each subpacket, The prediction residual is decoded with the number of bits based on the bit information, and prediction is performed for each subframe based on the decoded prediction residual, the leading sample value, and a prediction unit selected by the prediction unit selection information. Means for calculating a value;
From the calculated predicted value, the two-channel down-mix processing channel is acquired and the two-channel audio data obtained by the down-mixing is restored, and the multi-channel audio data is restored and restored. Means for converting 2-channel audio data and multi-channel audio data into an analog audio signal based on the sampling frequency and the number of quantization bits;
An audio signal receiving apparatus.
3) Correlation between two channels of downmix processing channels, which are channels obtained by correlating two channels obtained by downmixing three or more multi-channel audio signals, and a plurality of channels For each channel, the first sample value is obtained in units of frames for a predetermined time in response to the input audio signal, and a plurality of prediction units having different characteristics are used to obtain the current sample from the past in the time domain. A prediction unit that obtains a linear prediction value such that a prediction residual obtained from the predicted linear prediction value and the speech signal is minimized by further predicting a linear prediction value of each of the signals. Select by subframe and predictively encode,
The data structure includes a pack header including SCR information and user data including a compressed PCM access unit, and a plurality of the compressed PCM access units are provided in the frame. A first bitstream including the two-channel downmix processing channels arranged in the compressed PCM access unit, and prediction encoded data including prediction unit selection information indicating a prediction unit of each channel and a prediction residual; The predicted residual is packed and stored in a sub-packet composed of a second bit stream including other channels with the number of bits based on the designated bit information, and the data in the sub-packet is compressed with a variable bit rate. VBR identifier indicating that it is compressed data and the original analog on the playback side A synchronization information part including a sampling frequency and the number of quantization bits used for restoration to a voice signal is provided, and the head sample value is stored in the subpacket in the compressed PCM access unit. An audio signal transmission apparatus for packetizing and transmitting data encoded by the audio encoding method;
Receiving means for receiving packets packetized and transmitted by the audio signal transmission device;
Restoring means for restoring original data based on the received packet;
Time management means for managing the compressed PCM access unit of the restored data by SCR information;
Extracting a VBR identifier from the synchronization information part, taking a leading sample value from the subpacket based on the extracted identifier and taking out a prediction residual and a prediction part selection information indicating a prediction part from each subpacket, The prediction residual is decoded with the number of bits based on the bit information, and prediction is performed for each subframe based on the decoded prediction residual, the leading sample value, and a prediction unit selected by the prediction unit selection information. Means for calculating a value;
From the calculated predicted value, the two-channel down-mix processing channel is acquired and the two-channel audio data obtained by the down-mixing is restored, and the multi-channel audio data is restored and restored. Means for converting 2-channel audio data and multi-channel audio data into an analog audio signal based on the sampling frequency and the number of quantization bits;
An audio signal receiving device comprising:
An audio signal transmission system consisting of

  As described above, according to the present invention, for example, the original multi-channel audio signal is converted into a stereo 2-channel audio signal, and the obtained first group including the stereo 2-channel and other channels are included. Since it is classified into the second group and at least the audio signals of the channels of the second group are converted into correlated audio signals and each channel is predictively encoded, the multi-channel audio signals are predicted. In the case of encoding, the compression rate can be improved, and decoding can be performed using only the first group to reproduce two stereo channels.

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.

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 method Forward L, C, R like Dolby Surround method
3 channels + 1 channel in the rear S 4 channels in total (2) 5 channel system Like the Dolby AC-3 system SW channel, 3 channels in the front L, C, R + 2 channels in the rear SL, SR Total 5 channels (3) 6 channels system 6 channels (L, C, R, SW (Lfe), SL, SR like DTS (Digital Theater System) system and Dolby AC-3 system
)
(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

FIG. 1 shows a case where a multi-channel is compressed and playback side downmixing is prohibited as a transmission form of the first example. The 6-channel (ch) mix & matrix circuit 1 'on the encoding side is a front left (an example of a multi-channel signal).
Lf), center (C), front right (Rf), surround left (Ls),
6ch PCM with surround light (Rs) and Lfe (Low Frequency Effect)
The 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 ′. 2 and 3 described below
4, 5 and 6, the selection means 7 ′ is 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 includes the original 6ch (Lf, C, Rf, Ls, Rs, Lfe) and the coefficient mij (
i = 1, 2, j = 1, 2 to 6), the 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 respectively receive the first group channel “1”.
”,“ 2 ”and PCM data of the second group channels“ 3 ”to“ 6 ”are predictively encoded, and the 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”. Decode data into PCM data, then mix and matrix circuit 4 '
Based on the equations (1-2) and (2), the original 6ch (Lf, C, Rf, Ls, Rs
, Lfe) and the first group channels “1” and “2” are added,
Stereo 2ch data (L, R) is generated by subtraction.

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 there is no compression, the encoding side does not generate a correlation signal, and the original 6ch (Lf, C,
Rf, Ls, Rs, Lfe) PCM data is transmitted as it is (however, it is formatted), and after decoding on the decoding side, the original 6ch (Lf, C,
Rf, Ls, Rs, Lfe) only.

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 the expression (
The stereo 2ch data (L, R) is generated (downmixed) by 2).

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, on the encoding side, the following equation (1-
3) is converted into correlation signals for 6ch (1) to (6), and the encoding unit 2 ′ performs predictive encoding. On the decoding side, the original 6ch (Lf, C,
Rf, Ls, Rs, Lfe) only.
“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) by equation (2), and then the following equation (1)
-4) is converted into correlation signals “3” to “6” for the two channels “1” and “2” of the first group and the four channels of the second group as follows, and the first and second encodings are performed. Part 2 '
-1 and 2'-2 predictively encode each group channel. Then, on the decoding side, the original 6ch (Lf, C, Rf, Ls, Rs, Lf are obtained by the equations (1-4) and (2).
e) is restored and 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. Each channel “1”
The PCM data of “6” to “6” are stored in the 1-frame buffer 10 for each frame. Then, 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 each channel “1”.
The first sample data of each frame from “6” to “6” is applied to the formatting circuit 19. The prediction circuits 13D1, 13D2, and 15D1 to 15D4 respectively
For the PCM data of “1” to “6”, a plurality of linear prediction values of the current signal are calculated from past signals in the time domain by a plurality of predictors (not shown) having different characteristics,
Next, 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 stamps).
The minimum value of the prediction residual is selected for each subframe designated by the generator 17.

The selection signal / DTS generator 17 stores the bit number flag of the prediction residual in the packing circuit 1
8 and the formatting circuit 19, and a predictor selection flag indicating a predictor with the smallest prediction residual, a correlation coefficient a, and the decoding side uses the input buffer 22a (FIG. 1).
4) DTS indicating the time to extract stream data from 4) formatting circuit 1
9 is applied. The packing circuit 18 is a buffer / selector 14D1, 14D.
2, the prediction residual for 6ch selected by 16D1 to 16D4 is selected as a selection signal / DT
Packing is performed with the designated number of bits based on the bit number flag designated by the S generator 17. The PTS generator 17c is configured such that the decoding side uses the output buffer 110 (FIG. 14).
PTS (Presentation Time Stamp) indicating the time for extracting the PCM data from) is generated and output to the formatting circuit 19. The formatting circuit 19 also includes an encoding mode indicating compression / non-compression and downmix permission /
An identifier indicating prohibition is applied.

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, 6 channels, 7
A compression rate of 1% 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 is 4 for 2034 bytes of user data (A packet, V packet).
Byte pack start information and 6 byte SCR (System Clock Reference:
A system time base reference value) information, a 3-byte Mux rate information, and a 1-byte stuffing total 14-byte pack header are added (1 pack = total 2048 bytes). In this case, the time stamp SCR
By making the information continuous in the same title as “1” in the first pack, the time of the A pack in the same title can be managed.

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 Ar
ticle Number-International Standard Recording Code) number and UPC / E
AN-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 includes a sub-packet and a plurality of PPCMs as shown in FIG.
The PPCM access unit is composed of PPCM sync information and subpackets. 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. P after the second
The subpacket in the PCM access unit is composed of substream “0”, CRC, substream “1”, CRC and extra information except for the directory, and substreams “0” and “1” are restarted. It consists of a 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 depending on the sampling frequency fs
Or 160 is selected.
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 allocation information

The formatting circuit 19 also includes an ATSI (Audio / Audio / Audio / Audio) including management information as shown in FIGS. 12 and 13 for managing the audio packs shown in FIGS.
(Title Set Information). FIG.
OTT-AOB-ATR (audio only title / audio object set attribute) is indicated, and this AOTT-AOB-ATR (b127-
b0) is an audio encoding mode of 8 bits (b127 to b120) in order from the MSB side;
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 (b111 to b108) of channel group 1
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 quantization bit number Q1 of the channel group 1 is “0000b”, “0”
000b "
When the quantization bit number Q1 of the channel group 1 is “0001b”, “0”
000b "or" 0001b "
When the quantization bit number Q1 of the channel group 1 is “0010b”, “0”
000b "," 0001b "or" 0010b "
However, 0000b: 16 bits
0001b: 20 bits
0010b: 24 bits
Other: Hold

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

(5) Sampling frequency fs2 of channel group 2 (b99 to b96)
“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 for groups “1” and “2” from 1 channel (monaural) to 6 channels

FIG. 13 shows ATS-PG-CNT (audio title set / program / content), which is, in order from the top, 1 bit (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 *),
-4-bit (b19 to b16) downmix coefficient table number (DM-CO
EFTN)
・ Each RTI flags F15 to F of 1 bit each and 16 bits in total (b15 to b0)
0 is configured.
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 unit 3 ′ (3′-1, 3′-2) and the mix & matrix circuit 4
'Can also be realized by a computer program in addition to hardware. The variable rate bit stream data BS0, BS1 of the above format is
They are separated by the deformatting circuit 21. Then, the first sample data of one frame of each channel “1” to “6” and the predictor selection flag are set in the prediction circuit 24D
1, 24D2, 23D1 to 23D4, and the bit number flags of each channel “1” to “6” are applied to the unpacking circuit 22. The SCR, DTS, and prediction residual data string are applied to the input buffer 22a, and the PTS is output from the output buffer 110.
To be applied. An encoding mode indicating compression / non-compression and an identifier indicating permission / prohibition of downmixing 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 circuit 24
A plurality of predictors (not shown) in D1, 24D2, 23D1 to 23D4 are respectively
The same characteristics as those of the plurality of predictors in the encoding-side prediction circuits 13D1, 13D2, and 15D1 to 15D4, and the same characteristics are 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) seconds when fs = 96 kHz.
6. Variable length as shown in detail in FIG. The input buffer 22
The stream data stored in a 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 predicts the prediction circuits 24D1, 24D2, and 23D1, respectively.
To 23D4. 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. 1 each selected
The previous predicted value predicted by the two is added to calculate the current predicted 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. PC stored in output buffer 110
The M data is read out and output based on the PTS, and accordingly, FIG.
The variable-length access unit shown in FIG. 17 is expanded to output a fixed-length presentation unit shown in FIG.

Also, the sampling frequency fs and the quantization bit number Q in the PPCM sync information
Based on b, the PCM data is converted into an analog signal by the D / A converter 102. 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 As this search pointer, 1 second ahead, 1 second ahead, instead of 2 seconds ahead, 2
The one a second ago may be used.

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), and then the input buffer 2
Input / output control of 2a is performed (step S62), and then unpacking is performed (step S63). At this time, if there is a search reproduction instruction, the search pointer is decoded. The predictor is then selected based on the flag and decoded (
Next, input / output control of the output buffer 110 is performed (step S64).
65) Next, the original multi-channel is restored (step S66), then this is output (step S67), and this is repeated thereafter.

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 coding is performed, and the prediction coding data of the group having the highest compression rate is selected. For this reason, the encoding unit shown in FIG. 20 includes first to nth correlation circuits 1-1 to 1-n, and the n correlation circuits 1-1 to 1-n are, for example, 6
The PCM data of ch (Lf, C, Rf, Ls, Rs, Lfe) is converted into n types of 6ch signals “1” to “6” having different correlations.

For example, the first correlation circuit 1-1 converts as follows:
(1) = Lf
(2) = C− (Ls + Rs) / 2
(3) = Rf−Lf
(4) = Ls−a × Lfe
(5) = Rs−b × Rf
(6) = Lfe
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
To restore the original 6ch (Lf, C, Rf, Ls, Rs, Lfe),
Stereo 2ch data (L, R
) Is generated.

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;
Whether multi-channel data in the audio packet is compressed,
Alternatively, whether or not the multi-channel data in the audio packet is down-mixed in advance depending on whether the multi-channel data in the audio packet is allowed or prohibited to be down-mixed, or whether or not the down-mix coefficient is encoded. And a means for selecting
A speech encoding apparatus having the same.

1 '6ch mix & matrix circuit 13D1, 13D2, 15D1-15D4 Prediction circuit (buffer / selector 14)
The compression means is configured together with D1, 14D2, 16D1 to 16D4. )
14D1, 14D2, 16D1 to 16D4 Buffer / selector 17 Selection signal / DTS generator 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)

Obtained by correlating two or more multi-channel audio signals, which are obtained by down-mixing two channels obtained by down-mixing, and a plurality of channels. For each channel, the first sample value is obtained in units of frames for a predetermined time in response to the input audio signal, and the current signal from the past in the time domain is obtained by a plurality of prediction units having different characteristics. Sub-frames obtained by further dividing the frame into prediction units that obtain linear prediction values such that prediction residuals obtained from the predicted linear prediction values and the speech signal are minimized. Select units and predictively encode,
The data structure includes a pack header including SCR information and user data including a compressed PCM access unit, and a plurality of the compressed PCM access units are provided in the frame. A first bitstream including the two-channel downmix processing channels arranged in the compressed PCM access unit, and prediction encoded data including prediction unit selection information indicating a prediction unit of each channel and a prediction residual; The predicted residual is packed and stored in a sub-packet composed of a second bit stream including other channels with the number of bits based on the designated bit information, and the data in the sub-packet is compressed with a variable bit rate. VBR identifier indicating that it is compressed data and the original analog on the playback side A synchronization information part including a sampling frequency and the number of quantization bits used for restoration to a voice signal is provided, and the head sample value is stored in the subpacket in the compressed PCM access unit. An audio signal transmission apparatus for transmitting data encoded by an audio encoding method,
An audio signal transmission apparatus characterized in that the encoded encoded data including the head sample value, prediction unit selection information indicating the selected prediction unit, and prediction residual and the SCR information are packetized and transmitted. Obtained by correlating two or more multi-channel audio signals, which are obtained by down-mixing two channels obtained by down-mixing, and a plurality of channels. For each channel, the first sample value is obtained in units of frames for a predetermined time in response to the input audio signal, and the current signal from the past in the time domain is obtained by a plurality of prediction units having different characteristics. Sub-frames obtained by further dividing the frame into prediction units that obtain linear prediction values such that prediction residuals obtained from the predicted linear prediction values and the speech signal are minimized. Select units and predictively encode,
The data structure includes a pack header including SCR information and user data including a compressed PCM access unit, and a plurality of the compressed PCM access units are provided in the frame. A first bitstream including the two-channel downmix processing channels arranged in the compressed PCM access unit, and prediction encoded data including prediction unit selection information indicating a prediction unit of each channel and a prediction residual; The predicted residual is packed and stored in a sub-packet composed of a second bit stream including other channels with the number of bits based on the designated bit information, and the data in the sub-packet is compressed with a variable bit rate. VBR identifier indicating that it is compressed data and the original analog on the playback side A synchronization information part including a sampling frequency and the number of quantization bits used for restoration to a voice signal is provided, and the head sample value is stored in the subpacket in the compressed PCM access unit. Receiving means for receiving a packet in which data encoded by a voice encoding method is packetized and transmitted;
Restoring means for restoring original data based on the received packet;
Time management means for managing the compressed PCM access unit of the restored data by SCR information;
Extracting a VBR identifier from the synchronization information part, taking a leading sample value from the subpacket based on the extracted identifier and taking out a prediction residual and a prediction part selection information indicating a prediction part from each subpacket, The prediction residual is decoded with the number of bits based on the bit information, and prediction is performed for each subframe based on the decoded prediction residual, the leading sample value, and a prediction unit selected by the prediction unit selection information. Means for calculating a value;
From the calculated predicted value, the two-channel down-mix processing channel is acquired and the two-channel audio data obtained by the down-mixing is restored, and the multi-channel audio data is restored and restored. Means for converting 2-channel audio data and multi-channel audio data into an analog audio signal based on the sampling frequency and the number of quantization bits;
An audio signal receiving apparatus. Obtained by correlating two or more multi-channel audio signals, which are obtained by down-mixing two channels obtained by down-mixing, and a plurality of channels. For each channel, the first sample value is obtained in units of frames for a predetermined time in response to the input audio signal, and the current signal from the past in the time domain is obtained by a plurality of prediction units having different characteristics. Sub-frames obtained by further dividing the frame into prediction units that obtain linear prediction values such that prediction residuals obtained from the predicted linear prediction values and the speech signal are minimized. Select units and predictively encode,
The data structure includes a pack header including SCR information and user data including a compressed PCM access unit, and a plurality of the compressed PCM access units are provided in the frame. A first bitstream including the two-channel downmix processing channels arranged in the compressed PCM access unit, and prediction encoded data including prediction unit selection information indicating a prediction unit of each channel and a prediction residual; The predicted residual is packed and stored in a sub-packet composed of a second bit stream including other channels with the number of bits based on the designated bit information, and the data in the sub-packet is compressed with a variable bit rate. VBR identifier indicating that it is compressed data and the original analog on the playback side A synchronization information part including a sampling frequency and the number of quantization bits used for restoration to a voice signal is provided, and the head sample value is stored in the subpacket in the compressed PCM access unit. An audio signal transmission apparatus for packetizing and transmitting data encoded by the audio encoding method;
Receiving means for receiving packets packetized and transmitted by the audio signal transmission device;
Restoring means for restoring original data based on the received packet;
Time management means for managing the compressed PCM access unit of the restored data by SCR information;
Extracting a VBR identifier from the synchronization information part, taking a leading sample value from the subpacket based on the extracted identifier and taking out a prediction residual and a prediction part selection information indicating a prediction part from each subpacket, The prediction residual is decoded with the number of bits based on the bit information, and prediction is performed for each subframe based on the decoded prediction residual, the leading sample value, and a prediction unit selected by the prediction unit selection information. Means for calculating a value;
From the calculated predicted value, the two-channel down-mix processing channel is acquired and the two-channel audio data obtained by the down-mixing is restored, and the multi-channel audio data is restored and restored. Means for converting 2-channel audio data and multi-channel audio data into an analog audio signal based on the sampling frequency and the number of quantization bits;
An audio signal receiving device comprising:
An audio signal transmission system consisting of

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