JP2008282035A - Voice signal transmitter, voice signal receiver and voice signal transmission system - Google Patents

Abstract

<P>PROBLEM TO BE SOLVED: To provide a method for transmitting voice signals whose compressibility is improved by predictive coding. <P>SOLUTION: An adding circuit 1a computes the sum signal L+R of signals L and R of two stereophonic channels, and a subtracting circuit 1b computes the difference signal L-R. Difference arithmetic circuits 11D1 and 11D2 compute the current and last differences Δ(L+R) and Δ(L-R). Predictive coding circuits 15D1, 15D2, 16D1 and 16D2 compute a plurality of predicted values of the differences Δ(L+R) and Δ(L-R), compute respective predicted residues of the plurality of predicted values and the differences Δ(L+R) and Δ(L-R), and select the minimum predicted residue. Their predictive coded data is transmitted via a communication line. <P>COPYRIGHT: (C)2009,JPO&INPIT

Description

The present invention relates to an audio signal transmission apparatus, an audio signal reception apparatus, and an audio signal transmission system that are encoded by an audio encoding method for predictively encoding and compressing an audio signal.

As a method for predictive coding of a speech signal, the present inventor has disclosed an earlier application (Japanese Patent Application No. 9-289159).
), A plurality of linear prediction values of the current signal are calculated from past signals in the time domain by a plurality of predictors having different characteristics with respect to the original digital audio signal of one channel (channel), A method of calculating a prediction residual for each predictor from the plurality of linear prediction values and selecting a minimum value of the plurality of prediction residuals is proposed.

However, in the above method, the original digital audio signal has a sampling frequency of 96 kHz,
When the number of quantization bits is about 20 bits, a certain degree of compression effect can be obtained.
In recent DVD audio discs, twice the sampling frequency (= 192 kHz)
) Is used, and the number of quantization bits tends to be 24. Therefore, it is necessary to improve the compression rate.

Therefore, the present invention provides a transmission device, a speech signal reception device, and a speech signal transmission system for speech signals encoded by a speech coding method that can improve the compression rate when predictive coding of speech signals. With the goal.

In order to achieve the above object, the present invention comprises the following means 1) to 3).
That is,

1) The first and second audio signals of the same sampling frequency are subjected to matrix calculation and converted into two correlated channels that are correlated with each other,
A voice signal including the two converted correlation channels is obtained for each channel in response to an input voice signal, and a head sample value is obtained in units of a predetermined time frame, and time is determined by a plurality of linear prediction methods having different characteristics. The frame is further divided into a linear prediction method in which the linear prediction value of the current signal is predicted from the past of the region, and the prediction residual obtained from the predicted linear prediction value and the speech signal is minimized. Selected for each subframe, and predictive encoding,
The data structure includes header information including SCR information and user data including an audio compression PCM data portion, and the audio compression PCM data portion includes a plurality of access units. Speech that transmits data encoded by a speech coding method in which predictive coded data including a head sample value, a prediction residual, and a linear prediction method are stored in a subpacket arranged in the access unit. A signal transmission device,
An audio signal transmission apparatus characterized in that the selected head sample value, prediction residual, prediction encoded data including a linear prediction method, and the SCR information are packetized and transmitted through a communication line.
2) Matrix operation is performed on the first and second two audio signals having the same sampling frequency to convert them into two correlated channels, and
A voice signal including the two converted correlation channels is obtained for each channel in response to an input voice signal, and a head sample value is obtained in units of a predetermined time frame, and time is determined by a plurality of linear prediction methods having different characteristics. The frame is further divided into a linear prediction method in which the linear prediction value of the current signal is predicted from the past of the region, and the prediction residual obtained from the predicted linear prediction value and the speech signal is minimized. Selected for each subframe, and predictive encoding,
The data structure includes header information including SCR information and user data including an audio compression PCM data portion, and the audio compression PCM data portion includes a plurality of access units. Data encoded by a speech encoding method in which prediction encoded data including a head sample value, a prediction residual, and a linear prediction method are stored in a subpacket arranged in the access unit is packetized. Receiving means for receiving a packet transmitted via a communication line;
Restoring means for restoring original data based on the received packet;
Time management means for managing the audio compression PCM data part of the restored data by SCR information;
Means for calculating a predicted value from the selected leading sample value of the audio compressed PCM data portion, a linear prediction method, and a prediction residual;
Means for restoring the original audio signal from the calculated predicted value;
An audio signal receiving apparatus.
3) The first and second audio signals of the same sampling frequency are converted into two correlated channels that are correlated with each other by matrix calculation,
A voice signal including the two converted correlation channels is obtained for each channel in response to an input voice signal, and a head sample value is obtained in units of a predetermined time frame, and time is determined by a plurality of linear prediction methods having different characteristics. The frame is further divided into a linear prediction method in which the linear prediction value of the current signal is predicted from the past of the region, and the prediction residual obtained from the predicted linear prediction value and the speech signal is minimized. Selected for each subframe, and predictive encoding,
The data structure includes header information including SCR information and user data including an audio compression PCM data portion, and the audio compression PCM data portion includes a plurality of access units. Predictive encoded data including a head sample value, a prediction residual, and a linear prediction method are packetized from data encoded by a speech encoding method that is stored in a subpacket arranged in the access unit. An audio signal transmission device for transmitting via a communication line;
Receiving means for receiving packets packetized by the audio signal transmission apparatus and transmitted via the communication line;
Restoring means for restoring original data based on the received packet;
Time management means for managing the audio compression PCM data part of the restored data by SCR information;
Means for calculating a prediction value from the selected leading sample value, linear prediction method and prediction residual of the audio compression PCM data portion;
Means for restoring the original audio signal from the calculated predicted value;
An audio signal receiving device comprising:
An audio signal transmission system consisting of

As described above, according to the present invention, it is possible to transmit an audio signal whose compression rate is improved more than before and to decode the audio signal without any inconvenience.

Embodiments of the present invention will be described below with reference to the drawings. FIG. 1 shows a first example of a speech coding apparatus to which a speech signal transmission method according to the present invention is applied and a speech decoding apparatus corresponding thereto.
FIG. 2 is a block diagram showing the encoder of FIG. 1 in detail, FIG. 3 is an explanatory diagram showing the format of one frame multiplexed by the multiplexer of FIG. 2, and FIG. 4 is the format of the DVD pack FIG. 5 shows a DVD.
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 addition circuit 1a and a subtraction circuit 1b. In the adder circuit 1a, each channel (hereinafter, ch) is, for example, sampling frequency = 192 kHz
Then, the sum signal (L + R) of the stereo 2ch signals L and R with the number of quantization bits = 24 bits is calculated and output to the 1ch lossless encoder 2D1 for the sum channel, and the subtraction circuit 1b calculates the difference signal (LR). Then, it is output to the 1ch lossless encoder 2D2 for the difference channel. As shown in detail in FIG. 2, the encoders 2D1 and 2D2 are respectively sum signals (L + R).
The differences Δ (L + R) and Δ (LR) of the difference signal (LR) are predictively encoded and transmitted via a recording medium or a communication medium.

On the decoding side, as shown in detail in FIG. 6, the decoders 3D1 and 3D2 respectively decode the predicted encoded data of each channel into a sum signal (L + R) and a difference signal (LR),
Next, the channel correlation circuit B restores the sum signal (L + R) and the difference signal (LR) to the stereo 2ch signals L and R.

The encoders 2D1 and 2D2 will be described in detail with reference to FIG. Sum signal (
L + R) and the difference signal (LR) are stored in one frame buffer 10 for each frame. Then, each sample value (L + R), (LR) of one frame is applied to the difference calculation circuits 11D1, 11D2, respectively, and the difference Δ (L + R), Δ
(LR), that is, differential PCM (DPCM) data is calculated. In addition, the head sample values (L + R) and (LR) of each frame are applied to the multiplexer 19.

The difference Δ (L + R) calculated by the difference calculation circuit 11D1 is applied to a plurality of predictors 12a-1 to 12a-n and subtractors 13a-1 to 13a-n having different prediction coefficients. Each of the predictors 12a-1 to 12a-n calculates each prediction value of the difference Δ (L + R) based on each prediction coefficient, and each of the subtractors 13a-1 to 13b-n calculates each prediction value and the difference Δ. Each prediction residual of (L + R) is calculated. The buffer / selector 16D1 temporarily stores the plurality of prediction residuals, and the selection signal generator 17
The minimum prediction residual is selected for each subframe specified by the packing circuit 1
8 is output. Note that this subframe has a sample length of about one tens of frames, and one frame is 80 subframes as an example. Here, predictor 1
2a-1 to 12a-n and subtractors 13a-1 to 13a-n are prediction circuits 1 for the sum signal ch.
The prediction circuit 15D1 and the buffer / selector 16D1 form a prediction encoding circuit for the sum signal ch.

Similarly, the difference Δ (LR) calculated by the difference calculation circuit 11D2 is a plurality of predictors 12b-1 to 12b-n and subtractors 13b-1 to 13b- having different prediction coefficients.
applied to n. Then, in the predictors 12b-1 to 12b-n, respective predicted values of the difference Δ (LR) are calculated based on the respective prediction coefficients, and the subtractors 13b-1 to 13b.
In b−n, each prediction residual of each prediction value and difference Δ (LR) is calculated. The buffer / selector 16D2 temporarily stores the plurality of prediction residuals, selects the minimum prediction residual for each subframe specified by the selection signal generator 17, and outputs it to the packing circuit 18. Predictors 12b-1 to 12b-n and subtractors 13b-1 to 1
3b-n constitutes a prediction circuit 15D2 for the difference signal ch, and this prediction circuit 15D2
The buffer / selector 16D2 constitutes a predictive encoding circuit for the difference signal ch.

The selection signal generator 17 applies a prediction residual bit number flag (5 bits) to the packing circuit 18 and the multiplexer 19, and also predictor selection flags (the number n) indicating the predictor having the smallest prediction residual. 2 to 9 and 3 bits) is applied to the multiplexer 19. The packing circuit 18 includes buffers and selectors 16D1, 16
The prediction residuals for 2ch selected by D2 are packed with the designated number of bits based on the bit number flag designated by the selection signal generator 17.

The succeeding multiplexer 19 has a frame header (40 bits) for one frame as shown in FIG.
The first sample value (25 bits) of one frame of the sum signal ch (L + R),
The first sample value (25 bits) of one frame of the difference signal ch (LR),
Predictor selection flag for each subframe of the sum signal ch (L + R) (3 bits × 80
)When,
-Predictor selection flag (3 bits x 80) for each subframe of the difference signal ch (LR)
)When,
A bit number flag (5 bits × 80) for each subframe of the sum signal ch (L + R)
When,
-Bit number flag for each subframe of difference signal ch (LR) (5 bits x 80)
When,
A prediction residual data string (number of variable bits) of the sum signal ch (L + R);
-The prediction residual data string (number of variable bits) of the difference signal ch (LR) is 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 kHz, and the number of quantization bits =
In the case of 24 bits and 2 channels, a compression rate of 59% can be realized.

When this variable rate bit stream data is recorded on a DVD audio disk, it is packed into an audio (A) pack of compressed PCM shown in FIG. This pack consists of 2034 bytes of user data (A packet, V packet), 4 bytes of pack start information, and 6 bytes of SCR (System Cloning).
ck Reference: System time reference reference value) information and 3-byte Mux rate (rate
) A pack header of 14 bytes in total including information and 1 byte of stuffing is added (1 pack = total of 2048 bytes). In this case, the time of the A pack in the same title can be managed by setting the SCR information as a time stamp as “1” in the first pack in the ACB unit and continuing in the same title.

As shown in detail in FIG. 5, the compressed PCM A packet is composed of a 17, 9 or 14 byte packet header, a private header, and audio compressed PCM data of 1 to 2015 bytes in the format shown in FIG. Compression P
The CM 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;
-4 bytes of audio data information (ADI),
・ With stuffing byte of 0-7 bytes,
It is made up of.
Thus, the ADI of the compressed PCM A packet is selected to be 4 bytes, which is 4 bytes shorter than the ADI of the normal uncompressed PCM A packet. Therefore, the audio data can be increased by 4 bytes.

Next, the decoders 3D1 and 3D2 will be described with reference to FIG. The variable rate bit stream data in the format shown in FIG. 3 is separated by the demultiplexer 21 based on the frame header. Then, the first sample value of one frame of the sum signal ch (L + R) and the difference signal ch (LR) is the cumulative calculation circuit 2.
5a and 25b, the predictor selection flags of the sum signal ch (L + R) and the difference signal ch (LR) are the predictors (24a-1 to 24a-n) and (24b-1 to 24b-1), respectively.
24b-n) as a selection signal, a sum signal ch (L + R) and a difference signal c
The bit number flag of h (L−R) and the prediction residual data string are applied to the unpacking circuit 22. Here, predictors (24a-1 to 24a-n), (24b-1 to 24)
b-n) are encoding side predictors (12a-1 to 12a-n), (12b), respectively.
-1 to 12b-n), and the same characteristic is selected by the predictor selection flag.

The unpacking circuit 22 separates the prediction residual data string of the sum signal ch (L + R) and the difference signal ch (LR) based on each bit number flag, and adds each of the addition circuits 23.
a and 23b. In addition circuits 23a and 23b, current prediction residual data of sum signal ch (L + R) and difference signal ch (LR) from unpacking circuit 22, and predictors (24a-1 to 24a-n), (24b-1 to 24b-n)
Among them, the previous predicted value predicted by each one selected by the predictor selection flag is added to calculate the current predicted value. The predicted values this time are the differences Δ (L + R) and Δ (LR) calculated by the difference circuits 11a and 11b shown in FIG.
That is, DPCM data, and predictors (24a-1 to 24a-n), (24b
-1 to 24b-n) and the cumulative calculation circuits 25a and 25b.

Each of the cumulative calculation circuits 25a and 25b cumulatively adds the differences Δ (L + R) and Δ (LR) for each sample with respect to the first sample value of one frame, and adds the sum signal ch (L
+ R) and the PCM data of the difference signal ch (LR) are output. This sum signal (L +
As for R) and the difference signal (LR), a 2L signal is calculated by the adder circuit 4a and a 2R signal is calculated by the subtractor circuit 4b as shown in FIG. Then, the 2L signal and the 2R signal are respectively divided by 1/2 by the dividers 5a and 5b, 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, each difference Δ (L + R), Δ (LR) between the sum signal (L + R) and the difference signal (LR).
), That is, it is configured to predictively encode only DPCM data.

In the second embodiment, the sum signal (L + R), the difference signal (LR), that is, PCM data, or each difference Δ (L + R), Δ (LR), that is, DPCM data is selectively predictively encoded. It is configured as follows.

For this reason, in the encoding device shown in FIG. 7, the sum signal (L + R) is added to the configuration shown in FIG.
) And prediction circuits 15A and 15S for predictively encoding the difference signal (LR), respectively.
Buffer / selectors 16A and 16S are added. The selection signal generator 1
7 is a sum signal (L + R) selected by the buffers / selectors 16A and 16S.
), The difference signal (LR), and the minimum values of the respective prediction residuals of the differences Δ (L + R) and Δ (LR) selected by the buffers and selectors 16D1 and 16D2, respectively.
It is determined whether the PCM data or the DPCM data has a higher compression rate, and the higher data is selected. At this time, the PCM / DPCM selection flag (prediction circuit selection flag) 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 15 for the difference signal (LR).
When the prediction circuit 15D2 for S and the difference Δ (L−R) has the same configuration, the decoding apparatus does not need to provide both PCM data and DPCM data prediction circuits as shown in FIG. The prediction circuit may be used. Then, based on the prediction circuit selection flag transmitted from the encoding device, the selectors 26a and 26b select the outputs of the cumulative arithmetic circuits 25a and 25b in the case of DPCM data, and in the case of PCM data, the adder circuit 23a, Select the output of 23b.

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

For this reason, in the encoding device shown in FIG. 9, prediction circuits 15L and 15R for predictively encoding the original signals L and R and the buffer / selector 16 for the configuration shown in FIG.
L and 16R are added. The selection signal generator 17 is a buffer / selector 1.
Original signals L and R selected by 6L and 16R, and buffer / selectors 16A and 16
The sum signal (L + R) and difference signal (LR) selected by S and the respective residuals Δ (L + R) and Δ (LR) selected by the buffers / selectors 16D1 and 16D2 A group of data with a high compression rate is selected based on the minimum value. At this time, the selection flag (prediction circuit selection flag) is added and multiplexed.

In addition, when the three groups of prediction circuits shown in FIG. 9 have the same configuration, the decoding apparatus does not need to provide prediction circuits for three groups as shown in FIG. Based on the prediction circuit selection flag transmitted from the encoding device, the output of the cumulative arithmetic circuits 25a and 25b is selected in the case of DPCM data, and the output of the adder circuits 23a and 23b is selected in the case of PCM data. Then, the original signals L and R are restored by the channel correlation circuit B. Further, the selector 27a,
27b selects the output of the adder circuits 23a and 23b in the case of the group of the original signals L and R, and selects the output of the channel correlation circuit B in the other cases.

Also, when variable rate bitstream data predictively encoded by the encoding side is transmitted via the network, the encoding side packetizes it for transmission as shown in FIG. 11 (step S41), and then packet header (Step S42), and then the packet is sent out on the network (step S43).
). As shown in FIG. 12, 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).

FIG. 13 shows an aspect different from FIG. 3 of the audio (A) packet of the compressed PCM (PPCM) shown in FIG. In this different aspect, the audio data area in the compressed PCM (PPCM) audio (A) packet is composed of a plurality of PPCM access units as shown in FIG.
It consists of PCM sync information and subpackets. The subpackets in the first PPCM access unit are the directory, bitstream BS0, C
It is composed of RC and extra information, and the bit stream BS0 is composed of only PPCM blocks. Subpackets in the second and subsequent PPCM access units, except for the directory, are bitstream BS0, CRC,
The sub-stream BS0 at the head of the frame is composed of extra information, and is composed of a restart header and a PPCM block (including a frame head sample value).

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 restart header has information specifying the channel correlation circuit A (having an addition circuit and a subtraction circuit) for each frame. The variable rate bit stream data in the format shown in FIG. 13 is decoded into the original 2-channel audio signal by the decoders 3D1 and 3D2 having the configuration below the demultiplexer 21 in FIG.

FIG. 14 is a block diagram showing a second embodiment of the speech encoding apparatus and speech decoding apparatus according to the present invention. The channel correlation circuit A-1 shown in FIG. 14 includes an addition circuit 1a and a subtraction circuit 1b. The adding circuit 1a is a sum signal (L + of the stereo 2ch signals L and R).
R) is calculated, and the sum signal (L + R) is divided by ½ by the divider 5a, and then output to the lossless encoder 2D. The subtraction circuit 1b calculates the difference signal (LR). After dividing the difference signal (LR) by ½ by the divider 5b,
Output to the lossless encoder 2D. Lossless encoder 2D is 1/2 (
L + R) and 1/2 (L−R) are multiplexed to create 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) are obtained, and these are obtained from the channel correlation circuit B-
1 is supplied to an adder circuit 4a and a subtractor circuit 4b, respectively, and a stereo 2ch L signal and R signal are obtained as output signals. Note that a series of operations in the lossless encoder 2D and the lossless encoder 2D includes calculation of a difference, calculation of a prediction value, selection of a minimum prediction residual, calculation of a prediction value using the minimum prediction residual, and the like. This is performed in the same manner as in the embodiment. 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. Therefore, it can be decoded reliably. In addition, although the lossless compression for every frame was demonstrated to the example, it is not fixed and an area may be made into variable length.

1 is a block diagram showing a first embodiment of a speech encoding apparatus to which the present invention is applied and a speech decoding apparatus corresponding thereto. FIG. It is a block diagram which shows the encoder of FIG. 1 in detail. It is explanatory drawing which shows the format of 1 frame multiplexed by the multiplexer of 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 decoder of FIG. 1 in detail. It is a block diagram which shows the encoder of 2nd Embodiment. It is a block diagram which shows the decoder of 2nd Embodiment. It is a block diagram which shows the encoder of 3rd Embodiment. It is a block diagram which shows the decoder of 3rd Embodiment. It is a flowchart which shows the audio | voice transmission method. It is a flowchart which shows the audio | voice transmission method. FIG. 6 is a format explanatory diagram showing a mode different from FIG. 3 of the audio (A) packet of the compressed PCM (PPCM) shown in FIG. 5. It is a block diagram which shows 2nd Embodiment of the audio | voice encoding apparatus to which this invention is applied, and the audio | voice decoding apparatus corresponding to it.

Explanation of symbols

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 Predictors (the first predictive coding means is configured together with the subtractors 13a-1 to 13a-n and the buffer / selector 16D1)
12b-1 to 12b-n predictor (the second predictive encoding means is configured together with the subtracters 13b-1 to 13b-n and the buffer / selector 16D2)
13a-1 to 13a-n, 13b-1 to 13b-n Subtractors 16D1, 16D2, 16A, 16S, 16L, 16R Buffer / selector 15A Prediction circuit (the third predictive encoding means together with the buffer / selector 16A Constitute.)
15S prediction circuit (constitutes the fourth predictive encoding means together with the buffer / selector 16S)
15L prediction circuit (constitutes the fifth predictive encoding means together with the buffer / selector 16L)
15R prediction circuit (configures the sixth predictive encoding means together with the buffer / selector 16R)

Claims (3)

First and second audio signals of the same sampling frequency are converted into two correlated channels that are correlated with each other by matrix calculation,
A voice signal including the two converted correlation channels is obtained for each channel in response to an input voice signal, and a head sample value is obtained in units of a predetermined time frame, and time is determined by a plurality of linear prediction methods having different characteristics. The frame is further divided into a linear prediction method in which the linear prediction value of the current signal is predicted from the past of the region, and the prediction residual obtained from the predicted linear prediction value and the speech signal is minimized. Selected for each subframe, and predictive encoding,
The data structure includes header information including SCR information and user data including an audio compression PCM data portion, and the audio compression PCM data portion includes a plurality of access units. Speech that transmits data encoded by a speech coding method in which predictive coded data including a head sample value, a prediction residual, and a linear prediction method are stored in a subpacket arranged in the access unit. A signal transmission device,
An audio signal transmission apparatus characterized in that the selected head sample value, prediction residual, prediction encoded data including a linear prediction method, and the SCR information are packetized and transmitted through a communication line. First and second audio signals of the same sampling frequency are converted into two correlated channels that are correlated with each other by matrix calculation,
A voice signal including the two converted correlation channels is obtained for each channel in response to an input voice signal, and a head sample value is obtained in units of a predetermined time frame, and time is determined by a plurality of linear prediction methods having different characteristics. The frame is further divided into a linear prediction method in which the linear prediction value of the current signal is predicted from the past of the region, and the prediction residual obtained from the predicted linear prediction value and the speech signal is minimized. Selected for each subframe, and predictive encoding,
The data structure includes header information including SCR information and user data including an audio compression PCM data portion, and the audio compression PCM data portion includes a plurality of access units. Data encoded by a speech encoding method in which prediction encoded data including a head sample value, a prediction residual, and a linear prediction method are stored in a subpacket arranged in the access unit is packetized. Receiving means for receiving a packet transmitted via a communication line;
Restoring means for restoring original data based on the received packet;
Time management means for managing the audio compression PCM data part of the restored data by SCR information;
Means for calculating a predicted value from the selected leading sample value of the audio compressed PCM data portion, a linear prediction method, and a prediction residual;
Means for restoring the original audio signal from the calculated predicted value;
An audio signal receiving apparatus. First and second audio signals of the same sampling frequency are converted into two correlated channels that are correlated with each other by matrix calculation,
A voice signal including the two converted correlation channels is obtained for each channel in response to an input voice signal, and a head sample value is obtained in units of a predetermined time frame, and time is determined by a plurality of linear prediction methods having different characteristics. The frame is further divided into a linear prediction method in which the linear prediction value of the current signal is predicted from the past of the region, and the prediction residual obtained from the predicted linear prediction value and the speech signal is minimized. Selected for each subframe, and predictive encoding,
The data structure includes header information including SCR information and user data including an audio compression PCM data portion, and the audio compression PCM data portion includes a plurality of access units. Predictive encoded data including a head sample value, a prediction residual, and a linear prediction method are packetized from data encoded by a speech encoding method that is stored in a subpacket arranged in the access unit. An audio signal transmission device for transmitting via a communication line;
Receiving means for receiving packets packetized by the audio signal transmission apparatus and transmitted via the communication line;
Restoring means for restoring original data based on the received packet;
Time management means for managing the audio compression PCM data part of the restored data by SCR information;
Means for calculating a prediction value from the selected leading sample value, linear prediction method and prediction residual of the audio compression PCM data portion;
Means for restoring the original audio signal from the calculated predicted value;
An audio signal receiving device comprising:
An audio signal transmission system consisting of

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