JP3821382B2 - Optical recording medium and audio decoding device - Google Patents

Description

  The present invention relates to an optical recording medium and an audio decoding device for compressing a multi-channel audio signal with a variable length.

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, so the compression ratio needs to be improved. In addition, the sampling frequency and the number of quantization bits in multichannel may be different for each channel.

By the way, since a compression method such as the predictive coding method has a variable compression rate (VBR: variable bit rate), when a multi-channel audio signal is predictively encoded, the amount of data for each channel greatly changes over time. To do. When such data is transmitted, it is transmitted as a data stream instead of parallel for each channel.

Therefore, in order to enable reproduction (presentation) of such a variable length data stream in synchronization with each channel on the reproduction side (decoding side),
Decoding time indicating the timing for reading the data stream accumulated in the input buffer and outputting it to the decoder, and timing for reading the decoded data accumulated in the output buffer and outputting (presentation) to a speaker or the like You must manage playback time. Further, the playback side must manage the time for searching and playing back such a variable length data stream.

Therefore, an object of the present invention is to provide an optical recording medium and an audio decoding apparatus capable of managing the processing time on the reproducing side when encoding a multi-channel audio signal with a variable compression rate.

  In order to achieve the above object, the present invention comprises the following means 1) and 2).

That is,
And converting the stereo 2-channel audio signals by down-mix audio signals 1) original multi-channel,
By a predetermined matrix operation of the original multi-channel, the steps of the number of channels a correlation certain channel is converted to a correlation channel of the two channels fewer,
The audio signals of the stereo 2 channel and the correlation channel are obtained for each channel in response to the input audio signal, and a head sample value is obtained, and a plurality of linear prediction methods having different characteristics are used to obtain the current sample from the past in the time domain. Selecting and compressing a linear prediction method such that a linear prediction value of the signal is predicted, and a prediction residual obtained from the predicted linear prediction value and the speech signal is minimized;
Generating decoding time stamp information according to the amount of the compressed data;
The formatted packet is recorded by the step of formatting into a packet having user data including a packet header including the decoding time stamp information and compressed data, and the decoding time stamp The information is recorded as timing information for reading and decompressing compressed data that is separated from the user data and temporarily stored on the decoding side.
2) Downmix means for downmixing the original multi-channel audio signal and converting it into a stereo 2-channel audio signal;
Correlation means for converting the original multi-channel into a number of correlated channels, which are correlated channels and the number of channels is reduced by two channels, by a predetermined matrix operation;
The audio signals of the stereo 2 channel and the correlation channel are obtained for each channel in response to the input audio signal, and a head sample value is obtained, and a plurality of linear prediction methods having different characteristics are used to obtain the current sample from the past in the time domain. Compression means for selecting and compressing a linear prediction method in which a linear prediction value of a signal is predicted, and a prediction residual obtained from the predicted linear prediction value and the speech signal is minimized.
Timing generating means for generating decoding time stamp information indicating timing for reading compressed data in the input buffer on the decoding side according to the amount of data compressed by the compression means;
The original audio signal is decoded from the data recorded by the audio encoding device having the packet header including the decoding time stamp information and the means for formatting the packet with the user data including the compressed data. An audio decoding device,
Means for separating user data in the packet into a packet header and compressed data;
An input buffer for storing the separated compressed data;
Decoding means for reading out the compressed data stored in the input buffer based on decoding time stamp information in the packet header and restoring the stereo 2-channel audio signal and the original multi-channel audio signal;
A speech decoding apparatus having the same.

As described above, according to the present invention, since coding time stamp information is generated and set in the packet header according to the amount of compressed data, a multi-channel audio signal is encoded with a variable compression rate. In this case, the processing time on the playback side can be managed .

  Embodiments of the present invention will be described below with reference to the drawings. FIG. 1 is a block diagram showing a first embodiment of a voice encoding device to which the present invention is applied and a corresponding speech decoding device, FIG. 2 is a block diagram showing in detail the encoding unit of FIG. 1, and FIG. 1 is an explanatory diagram showing a bit stream encoded by the encoding unit of FIG. 2, FIG. 4 is an explanatory diagram showing a DVD pack format, FIG. 5 is an explanatory diagram showing a DVD audio pack format, and FIG. 1 is a block diagram showing in detail the decoding unit in FIG. 1, FIG. 7 is a timing chart showing the write / read timing of the input buffer in FIG. 6, FIG. 8 is an explanatory diagram showing the amount of compressed data for each access unit, and FIG. It is explanatory drawing which shows a presentation unit.

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 The 6-channel (ch) mix-and-matrix circuit 1 'on the control side 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 L
6ch PCM data of fe (Low Frequency Effect), 2ch “1” and “2” for the front group and 4ch “3” to “6” for the other groups according to the following equation (1)
2ch “1” and “2” are transferred to the first encoding unit 2′-1 and 4ch.
“3” to “6” are output to the second encoding unit 2′-2.

“1” = Lf + Rf
“2” = Lf−Rf
“3” = C− (Ls + Rs) / 2
“4” = Ls + Rs
“5” = Ls−Rs
“6” = Lfe−a × C
However, 0 ≦ a ≦ 1 (1)
As shown in detail in FIG. 2, the first and second encoding units 2′-1, 2′-2 constituting the encoding unit 2 ′ are respectively 2ch “1”, “2” and 4ch “3” to “3” to “3” to “3”. The PCM data of “6” is predictively encoded, and the 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 first and second decoding units 3′-1 and 3′-2 constituting the decoding unit 3 ′ perform 2ch “1” and “2” respectively related to the front group as shown in detail in FIG. The predictive encoded data of 4ch “3” to “6” related to other groups is decoded into PCM data.

Next, the original 6ch (Lf) based on the formula (1) by the mix & matrix circuit 4 ′
, C, Rf, Ls, Rs, Lfe), and the original 6ch and coefficient m
Stereo 2ch data (L, R) is generated as shown in the following equation (2) by ij (i = 1, 2, j = 1, 2 to 6).

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)
The encoding units 2′-1 and 2′-2 will be described in detail with reference to FIG. Each ch
The PCM data of “1” to “6” is 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 “
First sample data of each frame of “1” to “6” is applied to the formatting circuit 19. Each of the prediction circuits 13D1, 13D2, 15D1 to 15D4
A plurality of predictors (not shown) having different characteristics with respect to PCM data “1” to “6”
To calculate a plurality of linear prediction values of the current signal from past signals in the time domain, and then calculate a prediction residual for each predictor 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 stores the bit number flag of the prediction residual in the packing circuit 1
8 and the formatting circuit 19, the predictor selection flag indicating the predictor with the smallest prediction residual, the correlation coefficient a in the equation (1), and the decoding side is the input buffer 22a (FIG. 6). ) Is applied to the formatting circuit 19 indicating the time to extract the stream data from (). The packing circuit 18 is a buffer / selector 1.
The prediction residuals for 6ch selected by 4D1, 14D2, 16D1 to 16D4 are
Packing is performed 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. 6), and outputs the PTS to the formatting circuit 19.

The subsequent formatting circuit 19 formats the user data as shown in FIGS. The user data (subpacket) shown in FIG. 3 includes a variable rate bit stream (substream) BS0 including 2ch “1” and “2” predictive encoded data related to the forward group, and 4ch “3” to “3” related to other groups. A variable rate bitstream (substream) BS1 including predictive encoded data of “6”;
It consists of a bit stream 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",
・ Ch “6” coefficient a
Are 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. 2 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 detail in FIG. 5, 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 / 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. Both the forward access unit search pointer for searching the access unit after 1 second in the ADI and the backward access unit search pointer for searching for the access unit before 1 second are both 1 byte (specifically, Is set to the 7th byte of the ADI with the forward access unit search pointer and the 8th byte with the backward access unit search pointer.

Next, the decoding units 3′-1 and 3′-2 will be described with reference to FIG. 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 the prediction circuits 24D1, 24D2, respectively.
, 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 PTS is applied to the output buffer 110. Here, a plurality of predictors (not shown) in the prediction circuits 24D1, 24D2, and 23D1 to 23D4 are prediction circuits 13D1, 13D2, and 15D1-1 on the encoding side, respectively.
The characteristics are the same as those of the plurality of predictors in 5D4, and the same characteristics are selected by the predictor selection flag.

The stream data (predicted residual data string) separated by the deformatting circuit 21 is input to the input buffer 2 for each access unit by the SCR as shown in FIG.
2a is taken in and accumulated. Here, the data amount of one access unit is, for example, (1/96 kHz) seconds when fs = 96 kHz,
The length is variable as shown in detail in FIG. Then, the stream data stored in the input buffer 22a is read out by the FIFO based on the DTS and applied to the unpacking circuit 22.

The unpacking circuit 22 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 and output based on the PTS. Therefore, FIG. 9 (a)
The variable-length access unit shown in FIG. 9 is expanded, and a fixed-length presentation unit shown in FIG. 9B is output.

Here, when search reproduction is instructed via the operation unit 101, the control unit 10
The access unit is reproduced based on the forward access unit search pointer indicating 1 second ahead and the backward access unit search pointer indicating 1 second later placed in the ADI shown in FIG. 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 predictively encoded by the encoding units 2′-1 and 2′-2 shown in FIG. 2 is transmitted via the network, the encoding side uses the 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. 11A, 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. 11B, 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.

In the above embodiment, 2ch “1” and “2” related to the front group are set to “1” = Lf + Rf.
“2” = Lf−Rf
However, instead, the multi-channel is downmixed according to Equation (2) to generate stereo 2ch data (L, R), and then the following Equation (1) ′
“1” = L + R
“2” = LR
“3” to “5” are the same. “6” = Lfe-C (1) ′
(2nd embodiment). In this case, the decoding-side mix & matrix circuit 4 ′ can generate channel R by adding channels “1” and “2” and subtracting channel L by subtraction.

In addition, as shown in FIG. 12 as the third embodiment, stereo 2ch data (L
, R), and the stereo 2ch (L, R) and 4ch "3" to "6" may be predictively encoded. In the second and third embodiments, since the front left (Lf) and the front right (Rf) are not transmitted to the decoding side, they are generated by the equations (1) and (2) on the decoding side.

Next, a fourth 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. 13 includes first to n-th 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. 14, n correlation circuits 4-1 to 4-n (or coefficients a and b can be changed with respect to the correlation circuits 1-1 to 1-n on the encoding side. One correlation circuit (omitted) is provided. When the n groups of prediction circuits shown in FIG. 13 have the same configuration, the decoding device does not need to have n groups of prediction circuits as shown in FIG. 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.

It is a block diagram which shows 1st Embodiment of the audio | voice coding apparatus with which this invention is applied, and the audio | voice decoding apparatus corresponding to it. 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 a block diagram which shows the decoding part of FIG. 1 in detail. 7 is a timing chart showing write / read timings of the input buffer of FIG. 6. 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 3rd Embodiment of the audio | voice encoding apparatus with which this invention is applied, and the audio | voice decoding apparatus corresponding to it. It is a block diagram which shows the audio | voice coding apparatus of 4th Embodiment. It is a block diagram which shows the audio | voice decoding apparatus of 4th Embodiment.

Explanation of symbols

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 (timing generation means)
17c PTS generator (timing generator)
19 Formatting circuit (formatting means)
21 Deformatting circuit (separation means)
22 Unpacking circuit 22a Input buffer 24D1, 24D2, 23D1 to 23D4 Prediction circuit (expanding means)
100 Control unit (reading means)
110 Output buffer

Claims (2)

And converting the stereo 2-channel audio signals by down-mix audio signals of the original multi-channel,
By a predetermined matrix operation of the original multi-channel, the steps of the number of channels a correlation certain channel is converted to a correlation channel of the two channels fewer,
The audio signals of the stereo 2 channel and the correlation channel are obtained for each channel in response to the input audio signal, and a head sample value is obtained, and a plurality of linear prediction methods having different characteristics are used to obtain the current sample from the past in the time domain. Selecting and compressing a linear prediction method such that a linear prediction value of the signal is predicted, and a prediction residual obtained from the predicted linear prediction value and the speech signal is minimized;
Generating decoding time stamp information according to the amount of the compressed data;
The formatted packet is recorded by the step of formatting into a packet having user data including a packet header including the decoding time stamp information and compressed data, and the decoding time stamp The information is recorded as timing information for reading and decompressing compressed data that is separated from the user data and temporarily stored on the decoding side. Down-mix means for down-mixing the original multi-channel audio signal into a stereo 2-channel audio signal;
Correlation means for converting the original multi-channel into a number of correlated channels, which are correlated channels and the number of channels is reduced by two channels, by a predetermined matrix operation;
The audio signals of the stereo 2 channel and the correlation channel are obtained for each channel in response to the input audio signal, and a head sample value is obtained, and a plurality of linear prediction methods having different characteristics are used to obtain the current sample from the past in the time domain. Compression means for selecting and compressing a linear prediction method in which a linear prediction value of a signal is predicted, and a prediction residual obtained from the predicted linear prediction value and the speech signal is minimized.
Timing generating means for generating decoding time stamp information indicating timing for reading compressed data in the input buffer on the decoding side according to the amount of data compressed by the compression means;
The original audio signal is decoded from the data recorded by the audio encoding device having the packet header including the decoding time stamp information and the means for formatting the packet with the user data including the compressed data. An audio decoding device,
Means for separating user data in the packet into a packet header and compressed data;
An input buffer for storing the separated compressed data;
Decoding means for reading out the compressed data stored in the input buffer based on decoding time stamp information in the packet header and restoring the stereo 2-channel audio signal and the original multi-channel audio signal;
A speech decoding apparatus having the same.

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