JP2004139100A - Optical recording medium and voice decoding device - Google Patents

Abstract

<P>PROBLEM TO BE SOLVED: To manage a reproduction-side processing time when a multichannel voice signal is encoded with variable compression ratio. <P>SOLUTION: Predicting circuits 13D1, 13D2, and 15D1 to 15D4 and buffer selectors 14D1, 14D2, and 16D1 to 16D4 perform predictive encoding of a 6-channel voice signal. A DTS generator 17 generates decoding time stamp information showing the read timing of compressed data from a decoding-side input buffer 22a according to the amounts of predictively encoded data by channels and a formatting circuit 19 formats the data into a packet having a packet header including the decoding time stamp information and user data including the compressed data. <P>COPYRIGHT: (C)2004,JPO

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 disclosed a prior application (Japanese Patent Application No. 9-2980).
No. 89159), a plurality of predictors having different characteristics are used to calculate a plurality of linear prediction values of a current signal from a past signal in a time domain with respect to a one-channel original digital audio signal. A prediction coding method for calculating a prediction residual for each predictor from a plurality of linear prediction values and selecting a minimum value of the prediction residual has been proposed.

In the above method, the sampling frequency of the original digital audio signal is 96 kHz,
Although a certain degree of compression effect can be obtained when the quantization bit number is about 20 bits, a recent DVD audio disc has a sampling frequency (= 2 times) which is twice this.
192 kHz) and the number of quantization bits tends to be 24 bits. Therefore, it is necessary to improve the compression ratio. Further, the sampling frequency and the number of quantization bits in the multi-channel may be different for each channel.

By the way, the compression rate such as the predictive coding method has a variable compression rate (VBR: variable bit rate), and therefore, when predictive coding is performed on a multi-channel audio signal, the data amount of each channel greatly changes over time. I do. When transmitting such data, the data 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 on the reproduction side (decoding side) in synchronization with each channel,
A decoding time indicating the timing for reading the data stream stored in the input buffer and outputting the data to the decoder, and a timing for reading the decoded data stored in the output buffer and outputting (presenting) the data to a speaker or the like. Play time must be managed. In addition, 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 device that can manage the processing time on the reproduction side when encoding a multi-channel audio signal at a variable compression rate.

The present invention comprises means described in 1) and 2) below to achieve the above object.

That is,
1) down-mixing the original multi-channel audio signal and converting it into a stereo 2-channel audio signal;
Dividing into a first group including two stereo channels obtained by the step and a second group including correlated channels of the original multi-channel, at least the channels of the second group are the original channels By a predetermined matrix operation to convert the number of correlated channels into correlated channels whose number of channels is smaller by the two channels,
The audio signals of the stereo two channels and the correlated channels of the second group are obtained for each channel in response to an input audio signal, and a first sample value is obtained in a frame unit of a predetermined time, and a plurality of audio signals having different characteristics are obtained. A linear prediction method of predicting a linear prediction value of a current signal from the past in the time domain by the linear prediction method of the present invention, and minimizing a prediction residual obtained from the predicted linear prediction value and the audio signal. Selecting and predictively coding the frame in units of subframes further divided;
Grouping the prediction coded data including the linear prediction method selected in the step, the prediction residual, and a predetermined first sample value into a first group of the stereo two channels and a second group of the correlated channels. Storing in a divided bit stream;
Generating decoding time stamp information according to the amount of the compressed data;
Formatting the packet into packets having user data including the packet header including the decoding time stamp information and compressed data, wherein the formatted packet is recorded and the decoding time stamp An optical recording medium characterized in that information is recorded as timing information for reading and expanding compressed data separated from the user data and temporarily stored on the decoding side.
2) down-mixing the original multi-channel audio signal to convert it into a stereo 2-channel audio signal;
Dividing into a first group including two stereo channels obtained by the step and a second group including correlated channels of the original multi-channel, at least the channels of the second group are the original channels By a predetermined matrix operation to convert the number of correlated channels into correlated channels whose number of channels is smaller by the two channels,
The audio signals of the stereo two channels and the correlated channels of the second group are obtained for each channel in response to an input audio signal, and a first sample value is obtained in a frame unit of a predetermined time, and a plurality of audio signals having different characteristics are obtained. A linear prediction method of predicting a linear prediction value of a current signal from the past in the time domain by the linear prediction method of the present invention, and minimizing a prediction residual obtained from the predicted linear prediction value and the audio signal. Selecting and predictively coding the frame in units of subframes further divided;
Grouping the prediction coded data including the linear prediction method selected in the step, the prediction residual, and a predetermined first sample value into a first group of the stereo two channels and a second group of the correlated channels. Storing in a divided bit stream;
Generating decoding time stamp information according to the amount of the compressed data;
An audio decoding device for decoding an original audio signal from recorded data by formatting the packet into a packet having user data including the packet header including the decoding time stamp information and compressed data. hand,
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 accumulated in the input buffer based on the decoding time stamp information in the packet header and restoring the stereo two-channel audio signal and the original multi-channel audio signal;
Audio decoding device having the same.

As described above, according to the present invention, since the access unit search pointer is set in the packet header, the reproduction side can search and reproduce when encoding a multi-channel audio signal at a variable compression rate.

Hereinafter, embodiments of the present invention will be described with reference to the drawings. FIG. 1 is a block diagram showing a first embodiment of a voice encoding apparatus to which the present invention is applied and a speech decoding apparatus corresponding thereto, FIG. 2 is a block diagram showing the encoding unit of FIG. 1 in detail, and FIG. 1, 2 and 3. 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. 6 is an explanatory diagram showing a DVD audio pack format. FIG. 7 is a block diagram showing the decoding unit in detail in FIG. 1, FIG. 7 is a timing chart showing 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, FIG. 4 is an explanatory diagram showing a presentation unit.

Here, for example, the following four systems are known as the multi-channel system.
(1) 4 channel system L, C, R forward like Dolby surround system
3 channels + 1 channel of rear S, 4 channels in total (2) 5 channels system Like 3 channels of front L, C, R + 2 channels of rear SL, SR like the Dolby AC-3 system without SW channel 6 channels (L, C, R, SW (Lfe), SL, SR) such as DTS (Digital Theater System) and Dolby AC-3
)
(4) 8-channel system As in the case of the Sony Dynamic Digital Sound (SDDS) system, a total of 8 channels including 6 channels of front L, LC, C, RC, R, and SW + 2 channels of rear SL and SR are shown in FIG. The 6-channel (ch) mixing and matrix circuit 1 'on the conversion side includes front left (Lf), center (C), front right (Rf), surround left (Ls), and surround right (Rs) as examples of multi-channel signals. And L
The 6-ch PCM data of fe (Low Frequency Effect) is divided into 2ch “1” and “2” for the front group and 4ch “3” to “6” for the other groups by the following equation (1).
2ch “1” and “2” are assigned to the first encoding unit 2′-1 and 4ch
"3" to "6" are output to second encoding section 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 and 2'-2 that constitute the encoding unit 2 'respectively have 2ch "1", "2" and 4ch "3" to "4ch". The PCM data of "6" is predictively coded, and the predicted coded data is transmitted to the decoding side via the recording medium 5 and the communication medium 6 as a bit stream as shown in FIG. On the decoding side, the first and second decoding units 3′-1 and 3′-2 that constitute the decoding unit 3 ′ generate 2ch “1” and “2” for the front group as shown in detail in FIG. The predictive coded data of 4ch “3” to “6” relating to another group is decoded into PCM data.

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

L = m11 · Lf + m12 · Rf + m13 · C
+ M14 · Ls + m15 · Rs + m16 · Lfe
R = m21 · Lf + m22 · Rf + m23 · C
+ M24 · Ls + m25 · Rs + m26 · Lfe (2)
The encoding units 2'-1 and 2'-2 will be described in detail with reference to FIG. Each ch
The PCM data “1” to “6” are stored in the one-frame buffer 10 for each frame. Then, the sample data of each channel “1” to “6” of one frame is applied to the prediction circuits 13D1, 13D2, 15D1 to 15D4, respectively, and each channel “
The first sample data of each frame of “1” to “6” is applied to the formatting circuit 19. The prediction circuits 13D1, 13D2, 15D1 to 15D4 respectively
A plurality of predictors (not shown) having different characteristics for PCM data of “1” to “6”
Calculates a plurality of linear prediction values of the current signal from the past signal in the time domain, and then calculates prediction residuals for each predictor from the original PCM data and the plurality of linear prediction values. The following buffer / selectors 14D1, 14D2, 16D1 to 16D4 temporarily store the prediction residuals calculated by the prediction circuits 13D1, 13D2, 15D1 to 15D4, respectively, and provide a selection signal / DTS (decoding time stamp). The minimum value of the prediction residual is selected for each subframe specified by the generator 17.

The selection signal / DTS generator 17 stores the bit number flag of the prediction residual in the packing circuit 1.
8 to the formatting circuit 19, a predictor selection flag indicating the predictor with the smallest prediction residual, the correlation coefficient a in the equation (1), and the decoding side input buffer 22a (FIG. 6). ) Is applied to the formatting circuit 19, which indicates the time for extracting the stream data from the format circuit 19. The packing circuit 18 is a buffer / selector 1
The prediction residuals for 6 ch selected by 4D1, 14D2, 16D1 to 16D4 are
Packing is performed with the specified number of bits based on the bit number flag specified by the selection signal / DTS generator 17. Further, the PTS generator 17c generates a PTS (presentation time stamp) indicating a time at which the decoding side takes out the PCM data from the output buffer 110 (FIG. 6) and outputs the PTS to the formatting circuit 19.

The following formatting circuit 19 formats the user data as shown in FIGS. The user data (sub-packet) shown in FIG. 3 includes a variable-rate bit stream (sub-stream) BS0 including 2ch “1” and “2” prediction coded data for the front group, and 4ch “3” to 4ch for the other groups. A variable-rate bit stream (sub-stream) BS1 including the prediction encoded data of “6”;
It is composed of a bit stream header (restart header) provided before the substreams BS0 and BS1. Also, one frame of the substreams BS0 and BS1 has a frame header,
-First sample data of one frame of each channel "1" to "6";
A predictor selection flag for each subframe of each of the channels “1” to “6”;
A bit number flag for each subframe of each channel “1” to “6”;
A prediction residual data string (variable number of bits) for each channel “1” to “6”;
・ Coefficient a of ch “6”
Are multiplexed. According to such predictive coding, when the original signal has, for example, a sampling frequency = 96 kHz, the number of quantization bits = 24 bits, and 6 channels, 7
A compression ratio of 1% can be realized.

When the variable-rate bit stream data predictively encoded by the encoding units 2′-1 and 2′-2 shown in FIG. 2 is recorded on a DVD audio disc as an example of a recording medium, the audio data shown in FIG. A) Packed in a pack. This pack is 4 packs for 2034 bytes of user data (A packet, V packet).
Byte pack start information and 6-byte SCR (System Clock Reference:
It is configured by adding a 14-byte pack header of system byte reference value information, 3-byte Mux rate information, and 1-byte stuffing (1 pack = 2048 bytes in total). In this case, the time stamp SCR
By setting the information to be “1” in the first pack and being continuous within the same title, the time of the A pack within the same title can be managed.

As shown in detail in FIG. 5, the A packet of the compressed PCM is composed of a packet header of 19 or 14 bytes, a private header of the compressed PCM, and 1 to 2011 bytes of audio data (compressed PCM) in the format shown in FIG. ing. Then, the DTS and the PTS are set in the packet header of FIG. 5 (specifically, the PTS is set at the 10th to 14th bytes and the DTS is set at the 15th to 19th bytes). The compressed PCM private header is
A 1-byte substream ID,
-2-byte UPC / EAN-ISRC (Universal Product Code / European Ar
ticle Number-International Standard Recording Code) number and UPC / EAN-ISRC data,
A 1-byte private header length,
A 2 byte first access unit pointer;
8 bytes of audio data information (ADI);
・ With stuffing byte of 0-7 bytes,
It is composed of Both the forward access unit search pointer for searching for the access unit one second later and the backward access unit search pointer for searching for the access unit one second earlier in the ADI are both 1 byte (specifically, In the ADI, the forward access unit search pointer is set at the seventh byte and the backward access unit search pointer is set at the eighth byte.

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 head sample data of one frame of each of the channels “1” to “6” and the predictor selection flag are respectively stored in the prediction circuits 24D1 and 24D2.
, 23D1 to 23D4, and the bit number flags of the respective channels “1” to “6” are applied to the unpacking circuit 22. The SCR, the DTS, and the prediction residual data string are applied to the input buffer 22a, and the PTS is applied to the output buffer 110. Here, a plurality of predictors (not shown) in the prediction circuits 24D1, 24D2, 23D1 to 23D4 are respectively prediction circuits 13D1, 13D2, 15D1 to 1 on the encoding side.
The same characteristics as those of the plurality of predictors in 5D4, and those having the same characteristics are selected by the predictor selection flag.

The stream data (prediction 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 and is stored. Here, the data amount of one access unit is (1/96 kHz) seconds when fs = 96 kHz, for example.
The variable 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 each of the channels “1” to “6” based on each bit number flag, and separates the respective prediction circuits 24D1, 24D2, and 23D1.
To 23D4. The prediction circuits 24D1, 24D2, and 23D1 to 23D4 respectively use the current prediction residual data of each of the channels “1” to “6” from the unpacking circuit 22 and a predictor selection flag among a plurality of internal predictors. Each one selected
The current predicted value is calculated by adding the previous predicted value calculated by the first method, and then the PCM data of each sample is calculated based on the first sample data of one frame and stored in the output buffer 110. PC stored in output buffer 110
The M data is read and output based on the PTS. Therefore, FIG.
The variable length access unit shown in FIG. 9 is decompressed 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
0 reproduces the access unit based on the forward access unit search pointer indicating one second ahead and the backward access unit search pointer indicating one second later in the ADI shown in FIG. The search pointer may be one second ahead, two seconds ahead, two seconds ahead instead of one second ahead.

When variable-rate bit stream data predictively coded by the coding units 2′-1 and 2′-2 shown in FIG. 2 is transmitted via a network, the coding side performs transmission as shown in FIG. (Step S41), a packet header is added (step S42), and the packet is sent out to the network (step S43).

On the decoding side, as shown in FIG. 11A, the header is removed (step S51), the data is restored (step S52), and the data is stored in the memory and decoding is waited (step S53). Then, when decoding is performed, as shown in FIG. 11B, deformatting is performed (step S61).
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. Next, a predictor is selected and decoded based on the flag (
(Step S64) Then, input / output control of the output buffer 110 is performed (Step S64).
65), and then restore the original multi-channel (step S66), and then output this (step S67).

In the above embodiment, 2ch “1” and “2” for the front group are represented by “1” = Lf + Rf
“2” = Lf−Rf
, And performs predictive encoding. Instead, the multi-channel is downmixed by equation (2) to generate stereo 2-ch data (L, R), and then the following equation (1) ′
"1" = L + R
"2" = LR
“3” to “5” are the same “6” = Lfe−C (1) ′
May be used to perform predictive coding (second embodiment). In this case, the mix & matrix circuit 4 'on the decoding side can generate the channel L by adding the channels "1" and "2", and generate the channel R by subtracting the channel L.

As a third embodiment, as shown in FIG. 12, instead of 2ch "1" and "2", multi-channel is downmixed by equation (2) and stereo 2ch data (L
, R), and the stereo 2ch (L, R) and 4ch “3” to “6” may be predictively coded. In the second and third embodiments, since the front left (Lf) and the front right (Rf) are not transmitted to the decoding side, the decoding side generates them according to equations (1) and (2).

Next, a fourth embodiment will be described with reference to FIGS. In the above embodiment, one group of correlated signals "1" to "6" are configured to be predictively coded. In the fourth embodiment, a plurality of groups of correlated signals are generated. Then, it is configured to perform predictive coding and select predictive coded data of a group having the highest compression ratio. For this reason, the encoding unit shown in FIG. 13 is provided with first to n-th correlation circuits 1-1 to 1-n.
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 performs conversion as follows.

“1” = Lf + Rf
"2" = C-Lf
“3” = Rf−Lf
“4” = Ls−Lf
“5” = Rs−Lf
"6" = Lfe-C
A prediction circuit 15 and a buffer / selector 16 are provided for each of the correlation circuits 1-1 to 1-n, and a group having the highest compression rate is selected for correlation based on the data amount of the minimum value of the prediction residual for each group. Selected by the signal generator 17b. At this time, the formatting circuit 19 adds and multiplexes the selection flag (correlation circuit selection flag, correlation coefficients a and b of the correlation circuit).

Also, 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 correlated circuit omitted). When the prediction circuits of n groups shown in FIG. 13 have the same configuration, the decoding device does not need to provide the prediction circuits of n groups 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 6 ch (Lf, C, Rf, Ls, Rs, Lfe) are restored, and multi-channels are downmixed according to equation (2) to generate stereo 2-ch data (L, R).

In the first embodiment, one kind of correlated signals “1” to “6” is configured to be predictively coded. A group of original signals (Lf, C, Rf, Ls, Rs, Lfe) may be predictively coded and a group having a higher compression rate may be selected.

FIG. 1 is a block diagram illustrating a first embodiment of a speech encoding device to which the present invention is applied and a speech decoding device corresponding thereto. FIG. 2 is a block diagram illustrating an encoding unit of FIG. 1 in detail. FIG. 3 is an explanatory diagram illustrating a bit stream encoded by the encoding unit in FIGS. 1 and 2. FIG. 4 is an explanatory diagram showing a format of a DVD pack. FIG. 3 is an explanatory diagram showing a format of a DVD audio pack. FIG. 2 is a block diagram illustrating a decoding unit of FIG. 1 in detail. 7 is a timing chart showing write / read timings of the input buffer of FIG. FIG. 4 is an explanatory diagram showing a compressed data amount for each access unit. FIG. 3 is an explanatory diagram showing an access unit and a presentation unit. 5 is a flowchart illustrating a voice transmission method. 5 is a flowchart illustrating a voice transmission method. FIG. 11 is a block diagram showing a third embodiment of a speech encoding device to which the present invention is applied and a speech decoding device corresponding thereto. It is a block diagram showing a speech coding device of a fourth embodiment. It is a block diagram showing a speech decoding device of a fourth embodiment.

Explanation of reference numerals

1 '6ch Mix & Matrix Circuit 13D1, 13D2, 15D1-15D4 Prediction circuit (buffer / selector 14
D1, 14D2, 16D1 to 16D4 constitute compression means. )
14D1, 14D2, 16D1 to 16D4 Buffer / Selector 17 Selection Signal / DTS Generator (Timing Generation Means)
17c PTS generator (timing generation means)
19 Formatting circuit (formatting means)
21 Deformatting circuit (separation means)
22 Unpacking circuit 22a Input buffer 24D1, 24D2, 23D1 to 23D4 Prediction circuit (expansion means)
100 control unit (reading means)
110 output buffer

Claims (2)

Downmixing the original multi-channel audio signal into a stereo 2-channel audio signal;
Dividing into a first group including two stereo channels obtained by the step and a second group including correlated channels of the original multi-channel, at least the channels of the second group are the original channels By a predetermined matrix operation to convert the number of correlated channels into correlated channels whose number of channels is smaller by the two channels,
The audio signals of the stereo two channels and the correlated channels of the second group are obtained for each channel in response to an input audio signal, and a first sample value is obtained in a frame unit of a predetermined time, and a plurality of audio signals having different characteristics are obtained. A linear prediction method of predicting a linear prediction value of a current signal from the past in the time domain by the linear prediction method of the present invention, and minimizing a prediction residual obtained from the predicted linear prediction value and the audio signal. Selecting and predictively coding the frame in units of subframes further divided;
Grouping the prediction coded data including the linear prediction method selected in the step, the prediction residual, and a predetermined first sample value into a first group of the stereo two channels and a second group of the correlated channels. Storing in a divided bit stream;
Generating decoding time stamp information according to the amount of the compressed data;
Formatting the packet into packets having user data including the packet header including the decoding time stamp information and compressed data, wherein the formatted packet is recorded and the decoding time stamp An optical recording medium characterized in that information is recorded as timing information for reading and expanding compressed data separated from the user data and temporarily stored on the decoding side. Downmixing the original multi-channel audio signal into a stereo 2-channel audio signal;
Dividing into a first group including two stereo channels obtained by the step and a second group including correlated channels of the original multi-channel, at least the channels of the second group are the original channels By a predetermined matrix operation to convert the number of correlated channels into correlated channels whose number of channels is smaller by the two channels,
The audio signals of the stereo two channels and the correlated channels of the second group are obtained for each channel in response to an input audio signal, and a first sample value is obtained in a frame unit of a predetermined time, and a plurality of audio signals having different characteristics are obtained. A linear prediction method of predicting a linear prediction value of a current signal from the past in the time domain by the linear prediction method of the present invention, and minimizing a prediction residual obtained from the predicted linear prediction value and the audio signal. Selecting and predictively coding the frame in units of subframes further divided;
Grouping the prediction coded data including the linear prediction method selected in the step, the prediction residual, and a predetermined first sample value into a first group of the stereo two channels and a second group of the correlated channels. Storing in a divided bit stream;
Generating decoding time stamp information according to the amount of the compressed data;
An audio decoding device for decoding an original audio signal from recorded data by formatting the packet into a packet having user data including the packet header including the decoding time stamp information and compressed data. hand,
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 accumulated in the input buffer based on the decoding time stamp information in the packet header and restoring the stereo two-channel audio signal and the original multi-channel audio signal;
Audio decoding device having the same.

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