JP3387461B2 - Recording medium, audio decoding device - Google Patents

Description

Detailed Description of the Invention

[0001]

BACKGROUND OF THE INVENTION 1. Field of the Invention The present invention relates to a recording medium for recording a signal obtained by predictively encoding and compressing a voice signal, and a voice decoding.
Regarding the device .

[0002]

2. Description of the Related Art As a method of predictively encoding a voice signal,
The inventor of the present application (Japanese Patent Application No. 9-289159) uses a plurality of predictors having different characteristics for the original digital audio signal of one channel to convert the past signal from the past signal in the time domain to the present signal. Proposed a method of calculating a plurality of linear prediction values, calculating a prediction residual for each predictor from the original digital audio signal and the plurality of linear prediction values, and selecting the minimum value of the plurality of prediction residuals. There is.

[0003]

However, in the above method, the original digital audio signal has a sampling frequency = 96.
In the case of kHz and the number of quantization bits = about 20 bits, some compression effect can be obtained.
In an audio disc, a sampling frequency twice this (= 192 kHz) is used, and the number of quantization bits tends to be 24 bits, so that it is necessary to improve the compression rate.

SUMMARY OF THE INVENTION It is therefore an object of the present invention to provide a recording medium and a voice decoding device in which a signal having an improved compression rate is recorded when a voice signal is predictively coded.

[0005]

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

1) First and second of the same sampling frequency
A step of audio signals of two systems by matrix operation to convert into two correlation channels in correlation with each other, the stearate
Voice signal containing two correlated channels converted by
Signal for each channel in response to an input audio signal, and obtains a leading sample value in a frame unit for a predetermined time ,
Time-domain past using multiple linear prediction methods with different characteristics
To predict the linear predictor of the current signal from
A prediction obtained from the predicted linear prediction value and the speech signal.
Prediction Hakazansa selects the linear prediction method that minimizes
Encoding step, header information, compressed PCM access
User data including access unit and data structure including
And each of the chas selected in the above step.
Nell's linear prediction method, prediction residuals, and predetermined leading sample value
Predictive coded data including
The step stored in the subpacket placed in the unit
The predictive coded data is recorded by
The encoded data is used to recover the original speech signal.
Is recorded as data for calculating the predicted value
Recording medium characterized the Rukoto. 2) Means for expanding two compressed correlation signals which are data read from the recording medium according to claim 1, and decoding the audio signals of the first and second two systems by the expanded signals. speech decoding apparatus comprising: a means.

[0007]

BEST MODE FOR CARRYING OUT THE INVENTION Embodiments of the present invention will be described below with reference to the drawings. 1 is a block diagram showing a first embodiment of a speech coding apparatus and speech decoding apparatus applied to the present invention, FIG. 2 is a block diagram showing the encoder of FIG. 1 in detail, and FIG. 3 is multiplexing by the multiplexer of FIG. Become one
FIG. 4 is an explanatory diagram showing a frame format, 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 a block diagram showing the decoder of FIG. 1 in detail.

The channel correlation circuit A shown in FIG. 1 has an adding circuit 1a and a subtracting circuit 1b. Each channel (hereinafter, ch) of the adder circuit 1a has, for example, a sampling frequency = 192k.
The sum signal (L + R) of the stereo 2ch signals L and R of Hz and the number of quantization bits = 24 bits is calculated and output to the 1ch lossless encoder 2D1 for sum ch, and the subtraction circuit 1b outputs the difference signal (LR). It is calculated and output to the 1ch lossless encoder for difference channel 2D2. As shown in detail in FIG. 2, the encoders 2D1 and 2D2 perform predictive coding on the differences Δ (L + R) and Δ (L−R) of the sum signal (L + R) and the difference signal (LR), respectively, to thereby perform recording and communication media. To transmit through.

On the decoding side, as shown in detail in FIG. 6, the decoders 3D1 and 3D2 decode the predictive coded data of each channel into a sum signal (L + R) and a difference signal (LR), and then channel correlation. Circuit B uses this sum signal (L +
R) and the difference signal (LR) are restored to stereo 2ch signals L and R.

Referring to FIG. 2, encoders 2D1 and 2D2
Will be described in detail. The sum signal (L + R) and the difference signal (LR) are stored in the one frame buffer 10 for each frame. Then, each sample value (L +
R) and (LR) are the difference calculation circuits 11D1 and 1D1, 1
Applied to 1D2, the difference between this time and the previous time Δ (L + R), Δ
(LR), that is, the differential PCM (DPCM) data is calculated. Also, the leading sample value (L +
R) and (LR) are applied to the multiplexer 19.

The difference Δ (L + R) calculated by the difference calculation circuit 11D1 is calculated by a plurality of predictors 12 having different prediction coefficients.
a-1 to 12a-n and subtractors 13a-1 to 13a-n. Then, the predictors 12a-1 to 12a-n calculate the respective predicted values of the difference Δ (L + R) based on the respective prediction coefficients, and the subtractors 13a-1 to 13b-n respectively calculate the respective predicted values and the difference Δ. Each prediction residual of (L + R) is calculated. The buffer / selector 16D1 temporarily stores the plurality of prediction residuals, selects the minimum prediction residual for each subframe designated by the selection signal generator 17, and outputs the selected prediction residual to the packing circuit 18. Note that this subframe has a sample length of about one tenth of the frame, and one frame is 80 subframes as an example. here,
Predictors 12a-1 to 12a-n and subtractors 13a-1 to 1
3a-n form a prediction circuit 15D1 for the sum signal ch, and the prediction circuit 15D1 and the buffer / selector 16D1.
Constitutes a predictive coding circuit for the sum signal ch.

Similarly, the difference Δ (L−R) calculated by the difference calculation circuit 11D2 includes a plurality of predictors 12b-1 to 12b-n and subtracters 13b-1 to 13b having different prediction coefficients.
applied to b-n. Then, the predictors 12b-1 to 12b-12
In b-n, the difference Δ (L-
R) each predicted value is calculated, and the subtracters 13b-1 to 13b
In −n, the respective prediction residuals of the respective prediction values and the difference Δ (L−R) are calculated. The buffer / selector 16D2 temporarily stores the plurality of prediction residuals, and the selection signal generator 17
The minimum prediction residual is selected for each subframe designated by and is output to the packing circuit 18. Predictor 12b-
1 to 12b-n and subtractors 13b-1 to 13b-n form a prediction circuit 15D2 for the difference signal ch, and the prediction circuit 15D2 and the buffer / selector 16D2 form a prediction coding circuit for the difference signal ch. is doing.

The selection signal generator 17 applies the bit number flag (5 bits) of the prediction residual to the packing circuit 18 and the multiplexer 19, and the predictor selection flag (which indicates the predictor with the minimum prediction residual ( The number n is 2 to 9 and 3 bits are applied to the multiplexer 19. The packing circuit 18 is a buffer / selector 16D1, 16D.
The prediction residuals for 2ch selected by 2 are packed with the specified number of bits based on the bit number flag specified by the selection signal generator 17.

As shown in FIG. 3, the following multiplexer 19 is for one frame: a frame header (40 bits), a head sample value (25 bits) of one frame of the sum signal ch (L + R), and a difference. The first sample value (25 bits) of one frame of the signal ch (LR), the predictor selection flag (3 bits × 80) for each subframe of the sum signal ch (L + R), and the difference signal ch (L) -R) predictor selection flag (3 bits x 80) for each subframe; -bit number flag (5 bits x 80) for each subframe of sum signal ch (L + R);-difference signal ch (L- R) bit number flag (5 bits × 80) for each subframe, a prediction residual data string (variable number of bits) of the sum signal ch (L + R), and a prediction residual of the difference signal ch (L−R) Difference data string (variable number of bits ) And are multiplexed as an access unit and output as a variable rate bit stream. The prediction residual data string constitutes a subpacket. According to such predictive coding, the original signal is, for example, sampling frequency = 192.
In the case of kHz, the number of quantization bits = 24 bits, and two channels, a compression rate of 59% can be realized.

When recording the variable rate bit stream data on a DVD audio disc,
It is packed in the audio (A) pack of the compressed PCM shown in FIG. This pack has pack start information of 4 bytes for user data (A packet, V packet) of 2034 bytes and SCR (System Clock) of 6 bytes.
Reference: System time reference value) information, Mux rate information of 3 bytes, and stuffing of 1 byte, a pack header of 14 bytes in total is added (1 pack = 2048 bytes in total). In this case, the SCR information, which is a time stamp, is set to "1" in the first pack in the ACB unit and is consecutive in the same title, so that 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 packet header of 17, 9 or 14 bytes, a private header, and audio compressed PCM data of 1 to 2015 bytes in the format shown in FIG. ing. The private header of the compressed PCM includes: 1-byte substream ID, 2-byte UPC / EAN-ISRC (Universal Pr
oduct Code / European Article Number-International S
tandard Recording Code) number and UPC / EAN-
ISRC data, 1-byte private header length, 2-byte first access unit pointer, 4-byte audio data information (ADI), and 0-7 bytes of stuffing bytes. ing. In this way, AD of compressed PCM A packet
I is selected to be 4 bytes, and is the normal uncompressed PCM A
It is 4 bytes shorter than the ADI of the packet.
Therefore, the audio data can be increased by 4 bytes.

Next, referring to FIG. 6, decoders 3D1 and 3D
2 will be described. The variable rate bit stream data of the format shown in FIG.
1 separates based on the frame header. Then, the leading sample values of one frame of the sum signal ch (L + R) and the difference signal ch (L-R) are respectively calculated by the cumulative operation circuit 2
5a and 25b, and the predictor selection flags of the sum signal ch (L + R) and the difference signal ch (LR) are predictors (24a-1 to 24a-n) and (24b-1 to 24b), respectively.
-N) is applied as each selection signal and sum signal ch (L +
R) and the bit number flag of the difference signal ch (LR) and the prediction residual data string are applied to the unpacking circuit 22.
Here, the predictors (24a-1 to 24a-n), (24b
-1 to 24b-n) are respectively predictors (1
2a-1 to 12a-n), (12b-1 to 12b-n)
And the same characteristics are selected by the predictor selection flag.

The unpacking circuit 22 adds the sum signal ch (L
+ R) and the prediction residual data sequence of the difference signal ch (LR) are separated based on each bit number flag and output to the adder circuits 23a and 23b, respectively. In the adder circuits 23a and 23b, respectively, the sum signal ch from the unpacking circuit 22 is added.
(L + R) and the current prediction residual data of the difference signal ch (LR) and the predictors (24a-1 to 24a-n) and (24
b-1 to 24b-n), the previous predicted value predicted by each one selected by the predictor selection flag is added to calculate the current predicted value. This forecast value is
Differences Δ (L + R) and Δ (LR) calculated by the difference circuits 11a and 11b shown in FIG. 2, that is, DPC.
M data, predictors (24a-1 to 24a-n),
(24b-1 to 24b-n) and cumulative operation circuits 25a, 2
5b is applied.

The cumulative operation circuits 25a and 25b are respectively
Difference Δ (L +
R) and Δ (LR) are cumulatively added for each sample and PCM data of the sum signal ch (L + R) and the difference signal ch (LR) is output. The sum signal (L + R) and the difference signal (L-
As for R), the 2L signal is calculated by the adding circuit 4a and the 2R signal is calculated by the subtracting circuit 4b as shown in FIG. Then, the 2L signal and the 2R signal are divided into ½ by the dividers 5a and 5b, respectively, and the original stereo two-channel signals L and R are restored.

Next, a second embodiment will be described with reference to FIGS. In the above embodiment, the sum signal (L +
R), each difference Δ (L + R) of the difference signal (LR), Δ (L
-R), that is, it is configured to predictively encode only DPCM data, but in the second embodiment, a sum signal (L + R), a difference signal (LR), that is, PCM data, or each difference Δ thereof. (L + R), Δ (L−R), that is, DPCM data is selectively predictively coded.

Therefore, in the encoding device shown in FIG.
With respect to the configuration shown in (1), a sum signal (L + R) and a difference signal (L-
Prediction circuits 15A for predictively encoding R),
15S and buffer / selectors 16A and 16S are added. Further, the selection signal generator 17 is the buffer / selector 1.
Sum signal (L + selected by 6A and 16S, respectively)
R), difference signal (LR) and buffer / selector 16D
Differences Δ (L +
R), Δ (LR) based on the minimum value of each prediction residual,
It is determined which of the PCM data and the DPCM data has a higher compression rate, and the higher data is selected. At this time, the selection flag (prediction circuit selection flag) of the PCM / DPCM is added and multiplexed.

Here, the prediction circuit 15A for the sum signal (L + R) and the prediction circuit 15D1 for the difference Δ (L + R) shown in FIG. 7 have the same configuration, and the prediction circuit 15S for the difference signal (LR) is the same. When the prediction circuit 15D2 of the difference Δ (L−R) has the same configuration, the decoding device uses the PCM as shown in FIG.
It is not necessary to provide prediction circuits for both data and DPCM data, and a prediction circuit for one data is sufficient. Then, based on the prediction circuit selection flag transmitted from the encoder, the selectors 26a and 26b select the outputs of the cumulative operation circuits 25a and 25b in the case of DPCM data, and the PC
In the case of M data, the output of the adder circuits 23a and 23b is selected.

In the third embodiment, as shown in FIG. 9, original signals L and R (PCM data), sum signal (L + R), difference signal (LR) (PCM data), and their respective differences Δ. (L
+ R), Δ (LR) (DPCM data), one of three groups is selectively predictively coded.

Therefore, in the encoding device shown in FIG.
Prediction circuits 15L and 15R and buffer / selectors 16L and 16R for predictively encoding the original signals L and R, respectively, are added to the configuration shown in FIG. The selection signal generator 17 includes the original signals L and R selected by the buffer / selectors 16L and 16R and the buffer / selectors 16A and 16S.
Sum signal (L + R), difference signal (LR) selected by
And the data of the group having a high compression rate based on the minimum value of the prediction residuals of the differences Δ (L + R) and Δ (L−R) selected by the buffer / selectors 16D1 and 16D2. At this time, the selection flag (prediction circuit selection flag) is added and multiplexed.

When the three groups of prediction circuits shown in FIG. 9 have the same structure, the decoding device does not need to have three groups of prediction circuits as shown in FIG. 10, and one group of prediction circuits is not necessary. Good. Then, based on the prediction circuit selection flag transmitted from the encoding device, the output of the cumulative operation circuits 25a and 25b is selected in the case of DPCM data, and the addition circuits 23a and 23b in the case of PCM data.
Of the original signal L by the channel correlation circuit B,
Restore R. Then, in the case of the group of the original signals L and R, the selectors 27a and 27b add the adder circuit 23.
The outputs of a and 23b are selected, and in other cases, the output of the channel correlation circuit B is selected.

When the variable rate bit stream data predictively coded by the encoding side is transmitted through the network, the encoding side packetizes it for transmission as shown in FIG. 11 (step S41). Next, a packet header is added (step S42), and then this packet is sent out on the network (step S4).
3). On the decoding side, the header is removed as shown in FIG. 12 (step S51), the data is then restored (step S52), this data is then stored in the memory and awaits decoding (step S53).

FIG. 13 shows a mode different from that of the audio (A) packet of the compressed PCM (PPCM) shown in FIG. In this different aspect, the audio data area in the audio (A) packet of compressed PCM (PPCM) is composed of a plurality of PPCM access units as shown in FIG.
It is composed of CM sync information and subpackets.
The subpacket in the first PPCM access unit is
Directory, bitstream BS0, CRC,
Bitstream BS composed of extra information
0 is composed of only PPCM blocks. The sub-packets in the second and subsequent PPCM access units are the bit stream BS0 except the directory,
The substream BS0 at the beginning of the frame, which is composed of CRC and extra information, has a restart header and PPC.
It is composed of M blocks (including the sample value at the beginning of the frame).

The PPCM sync information (hereinafter also referred to as sync information) includes the following information. Number of samples per packet: 40, 80 or 160 is selected according to the sampling frequency fs. -Data rate: "0" in the case of VBR (identifier indicating that the data in the subpacket is compressed data) -Sampling frequency fs and quantization bit number Qb-Channel allocation information Restart header is a channel for each frame Correlation circuit A
It has information clearly specifying (having an addition circuit and a subtraction circuit). The variable rate bit stream data of the format shown in FIG. 13 is the demultiplexer 2 of FIG.
The original 3 channel audio signal is decoded by the decoders 3D1 and 3D2 having a configuration of 1 or less.

FIG. 14 is a block diagram showing a second embodiment of a speech coding apparatus and speech decoding apparatus according to the present invention. The channel correlation circuit A-1 shown in FIG.
It has a and a subtraction circuit 1b. Adder circuit 1a is stereo 2
The sum signal (L + R) of the ch signals L and R is calculated, the sum signal (L + R) is divided by 1/2 by the divider 5a, and then output to the lossless encoder 2D, and the subtraction circuit 1b.
Calculates the difference signal (LR), divides this difference signal (LR) by 1/2 by the divider 5b, and then outputs it to the lossless encoder 2D. Lossless encoder 2
D uses 1/2 (L + R) and 1/2 (L-R) to multiplex them to form a multiplexed signal 250. The multiplexed signal 250 is decoded by the lossless decoder 3D to obtain the original 1/2 (L + R) and 1/2 (LR), which are added by the adder circuit 4a forming the channel correlation circuit B-1. To the subtraction circuit 4b, and the stereo 2ch L and R signals are obtained as output signals. It should be noted that the lossless encoder 2D and a series of operations in the lossless encoder 2D, such as difference calculation, prediction value calculation, minimum prediction residual difference selection, and prediction value calculation using the minimum prediction residual error, are performed by the first method. It is performed in the same manner as the embodiment of.
The variable rate bit stream data in the format shown in FIG. 13 is identified by the identifier stored in the restart header of the PPCM access unit, for example, whether the channel correlation circuit of FIG. 1 or the channel correlation circuit of FIG. 14 is used. Since it is set, it can be surely decoded in any case. Note that the lossless compression for each frame has been described as an example, but the section may have a variable length instead of being fixed.

[0030]

As described above, according to the present invention, in particular, two correlation signals calculated by the channel correlation circuit are obtained.
In response to the audio signal input for each channel.
Sample value and predict from past signal in time domain
The prediction error among the multiple predictions of the current signal
Lossless compression using the linear prediction method with the minimum value
Because it was cow, the compression ratio in the case of predictive coding an audio signal
It is possible to provide a recording medium that improves the above and a decoding device for the recording medium.

[Brief description of drawings]

FIG. 1 is a block diagram showing a first embodiment of a speech coding apparatus and speech decoding apparatus applied to the present invention.

FIG. 2 is a detailed block diagram of the encoder of FIG.

FIG. 3 is an explanatory diagram showing a format of one frame multiplexed by the multiplexer of FIG.

FIG. 4 is an explanatory diagram showing a format of a DVD pack.

FIG. 5 is an explanatory diagram showing a format of a DVD audio pack.

6 is a detailed block diagram of the decoder of FIG. 1. FIG.

FIG. 7 is a block diagram showing an encoder of a second embodiment.

FIG. 8 is a block diagram showing a decoder of the second embodiment.

FIG. 9 is a block diagram showing an encoder of a third embodiment.

FIG. 10 is a block diagram showing a decoder according to a third embodiment.

FIG. 11 is a flowchart showing a voice transmission method.

FIG. 12 is a flowchart showing a voice transmission method.

13 is a format explanatory diagram showing an aspect different from that of FIG. 3 of the audio (A) packet of the compressed PCM (PPCM) shown in FIG. 5.

FIG. 14 is a block diagram showing a second embodiment of a speech coding apparatus and speech decoding apparatus applied to the present invention.

[Description of Reference Signs] 1a, 4a Addition circuit (addition means) 1b, 4b Subtraction circuit (subtraction means) 5a, 5b Divider 11D1 Difference calculation circuit (first difference calculation means) 11D2 Difference calculation circuit (second difference calculation) Means) 12a-1 to 12a-n Predictor (subtractor 13a-1 to
13a-n and the buffer / selector 16D1 constitute a first predictive coding means. ) 12b-1 to 12b-n Predictor (subtractor 13b-1 to
13b-n and the buffer / selector 16D2 constitute second predictive coding means. ) 13a-1 to 13a-n, 13b-1 to 13b-n Subtractors 16D1, 16D2, 16A, 16S, 16L, 16R
Buffer / selector 15A Prediction circuit (with buffer / selector 16A
The predictive coding means of is constructed. ) 15S Prediction circuit (fourth with buffer / selector 16S)
The predictive coding means of is constructed. ) 15L prediction circuit (fifth with buffer / selector 16L)
The predictive coding means of is constructed. ) 15R prediction circuit (sixth with buffer / selector 16R)
The predictive coding means of is constructed. )

─────────────────────────────────────────────────── ─── Continuation of the front page (56) References JP-A-64-44499 (JP, A) JP-A-2-127899 (JP, A) JP-A-10-233058 (JP, A) JP-A-10- 320928 (JP, A) (58) Fields investigated (Int.Cl. ⁷ , DB name) G10L 19/00

Claims (2)

(57) [Claims] 1. A first and a second of the same sampling frequency
Converting the audio signals of the two systems to two correlated channels in correlation with each other matrix operation, the two correlation channel converted by the step free
For each channel, the leading sample value is obtained in frame units for a predetermined time in response to the input audio signal for each channel, and the time range is also measured by a plurality of linear prediction methods with different characteristics.
The linear predicted values of the past and present signals of
Obtained from the predicted linear prediction value and the speech signal.
A predictive coding method that minimizes the predicted residual error , predictive-encodes the header, and includes a header information and a compressed PCM access unit.
The data structure including the data and
Linear prediction method and prediction for each channel selected by step
Prediction-coded data that includes the measurement residual and a predetermined leading sample value
Are placed in the compressed PCM access unit.
The step of storing in the predictive encoded data is recorded by the step of storing in the predictive encoded data.
Is the predicted value used to recover the original speech signal.
Characterized by being recorded as data for calculation
Recording medium shall be the. 2. A means for expanding two compressed correlation signals, which are data read from the recording medium according to claim 1, and the two expanded audio signals of the first and second systems by the expanded signals. speech decoding apparatus having a means for decoding.