JP2004326136A - Voice signal transmitting method and voice decoding method - Google Patents

Abstract

<P>PROBLEM TO BE SOLVED: To provide a voice signal transmitting method and a voice decoding method that can manage a reproduction side processing time when a multi-channel voice signal is encoded with variable compressibility. <P>SOLUTION: Predicting circuits 13D1 and 13D2, and 15D1 to 15D4, and buffer selectors 14D1, 14D2, and 16D1 to 16D4 predict and encode a 6-channel voice signal. A DTS generator 17 generates decoding time stamp information indicating the timing of readout of compressed data in an input buffer 22a on a decoding side according to the amounts of predictively encoded data by channels and a formatting circuit 19 formats data into a packet having a packet header including the decoding time stamp information and user data including compressed data and transmits the formatted data through a communication line. <P>COPYRIGHT: (C)2005,JPO&NCIPI

Description

  The present invention relates to a method for transmitting a multi-channel audio signal via a communication line and an audio decoding method.

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 audio signal transmission method and an audio decoding method that can manage the processing time on the reproduction side when encoding a multi-channel audio signal at a variable compression ratio.

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

1) A multi-channel audio signal is obtained in response to an input audio signal for each channel as it is or for each channel correlated with each other, a first sample value is obtained, and a time domain is obtained by a plurality of linear prediction methods having different characteristics. , A linear prediction value of the current signal is predicted from the past signal, and a linear prediction method that minimizes the prediction residual obtained from the predicted linear prediction value and the audio signal is selected and compressed. Steps and
Generating decoding time stamp information indicating the timing of reading compressed data in the input buffer on the decoding side according to the amount of the compressed data;
A data structure including user data including a packet header including the decoding time stamp information, a compressed PCM private header and the compressed data, and a UPC / EAN-ISRC number and a UPC / An audio signal transmission method, wherein EAN-ISRC data is formatted into a packet to be placed in the compressed PCM private header, and the formatted user data is transmitted in a predetermined transmission format.
2) In response to an input audio signal for each channel as it is or for each channel correlated with each other, a multi-channel audio signal is obtained in the time domain by a plurality of linear prediction methods having different characteristics while obtaining a leading sample value. , A linear prediction value of the current signal is predicted from the past signal, and a linear prediction method that minimizes the prediction residual obtained from the predicted linear prediction value and the audio signal is selected and compressed. Steps and
Generating decoding time stamp information indicating the timing of reading compressed data in the input buffer on the decoding side according to the amount of the compressed data;
A data structure including user data including a packet header including the decoding time stamp information, a compressed PCM private header and the compressed data, and a UPC / EAN-ISRC number and a UPC / An audio decoding method for decoding an original audio signal from data formatted into packets in which EAN-ISRC data is placed in the compressed PCM private header,
Calculating a prediction value from the compressed data including the selected leading sample value, prediction residual, and linear prediction method;
Restoring the multi-channel audio signal from the calculated predicted value;
An audio decoding method comprising:

As described above, according to the present invention, decoding time stamp information indicating the timing of reading compressed data from the input buffer on the decoding side is generated and set in the packet header according to the amount of compressed data for each channel. Therefore, when encoding a multi-channel audio signal with a variable compression ratio, the processing time on the reproduction side can be managed.

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 speech encoding apparatus according to the present invention and a speech decoding apparatus corresponding to the speech encoding apparatus. FIG. 2 is a block diagram showing the encoding unit of FIG. 1 in detail. 1 and 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. 6 is a decoding diagram of FIG. FIG. 7 is a block diagram specifically showing the conversion unit, FIG. 7 is a timing chart showing write / read timings 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. 9 is an explanatory diagram showing an access unit and 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

The 6-channel (ch) mix & matrix circuit 1 'on the encoding side shown in FIG. 1 includes front left (Lf), center (C), front right (Rf), surround left (Ls), Surround light (Rs) 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 which 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 or 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 constituting the decoding unit 3 'respectively provide 2ch "1" and "2" for the front group as shown in detail in FIG. The predictive coded data of 4ch “3” to “6” for other groups 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 / E
AN-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 applied to the present invention 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. It is a block diagram which shows the audio | voice encoding device of 3rd Embodiment, and the audio | voice decoding device corresponding to it. It is a block diagram showing a speech coding device of a fourth embodiment. It is a block diagram showing the speech decoding device of a 4th 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)

A multi-channel audio signal is obtained for each channel as it is or for each channel correlated with each other, in response to the input audio signal, a first sample value is obtained, and the past in the time domain is obtained by a plurality of linear prediction methods having different characteristics. A linear prediction value of the current signal is predicted from the signals of the signals, and selecting and compressing a linear prediction method that minimizes a prediction residual obtained from the predicted linear prediction value and the audio signal; ,
Generating decoding time stamp information indicating the timing of reading compressed data in the input buffer on the decoding side according to the amount of the compressed data;
A data structure including user data including a packet header including the decoding time stamp information, a compressed PCM private header and the compressed data, and a UPC / EAN-ISRC number and a UPC / An audio signal transmission method, wherein EAN-ISRC data is formatted into a packet to be placed in the compressed PCM private header, and the formatted user data is transmitted in a predetermined transmission format. A multi-channel audio signal is obtained for each channel as it is or for each channel correlated with each other, in response to the input audio signal, a first sample value is obtained, and the past in the time domain is obtained by a plurality of linear prediction methods having different characteristics. A linear prediction value of the current signal is predicted from the signals of the signals, and selecting and compressing a linear prediction method that minimizes a prediction residual obtained from the predicted linear prediction value and the audio signal; ,
Generating decoding time stamp information indicating the timing of reading compressed data in the input buffer on the decoding side according to the amount of the compressed data;
A data structure including user data including a packet header including the decoding time stamp information, a compressed PCM private header and the compressed data, and a UPC / EAN-ISRC number and a UPC / An audio decoding method for decoding an original audio signal from data formatted into packets in which EAN-ISRC data is placed in the compressed PCM private header,
Calculating a prediction value from the compressed data including the selected leading sample value, prediction residual, and linear prediction method;
Restoring the multi-channel audio signal from the calculated predicted value;
An audio decoding method comprising:

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