JP2001175291A - Sound encoding method - Google Patents

Abstract

PROBLEM TO BE SOLVED: To manage the processing time of a reproduction side when the sound signal of a multi-channel is encoded by variable compressibility. SOLUTION: Prediction circuits 31D1, 31D2, 15D1 to 15D4 and buffer selectors 14D1, 14D2 and 16D1 to 16D4 predictively encode 6 channel sound signals. A DTS generator 17 generate decoding time stamp information showing timing for reading compression data in the input buffer 22a of a decoding side in accordance with prediction/encoding data quantity for respective channels. A formatting circuit 19 formats a packet header including decoding time stamp information and a user data having compression data into a packet.

Description

DETAILED DESCRIPTION OF THE INVENTION

[0001]

BACKGROUND OF THE INVENTION 1. Field of the Invention The present invention relates to an audio encoding method for compressing a multi-channel audio signal with a variable length.

[0002]

2. Description of the Related Art As a method of compressing an audio signal with a variable length, the present inventor has filed a prior application (Japanese Patent Application No. 9-289159).
Calculates a plurality of linear prediction values of a current signal from a past signal in the time domain by using a plurality of predictors having different characteristics with respect to the one-channel original digital audio signal. A prediction encoding method for calculating a prediction residual for each predictor from a prediction value and selecting a minimum value of the prediction residual has been proposed.

In the above method, the original digital audio signal has a sampling frequency = 96 kHz and the number of quantization bits = 2.
Although a certain degree of compression effect can be obtained in the case of about 0 bits, in recent DVD audio discs, this 2
Since a double sampling frequency (= 192 kHz) is used and the number of quantization bits tends to be 24 bits, 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.

[0004]

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

Therefore, in order to allow such a variable-length data stream to be reproduced (presented) on the reproduction side (decoding side) in synchronization with each channel, the data stream stored in the input buffer is read out and the decoder is read out. It is necessary to manage a decoding time indicating a timing for outputting the data to the output buffer and a reproduction time indicating a timing for reading out the decoded data stored in the output buffer and outputting (presenting) the data to a speaker or the like. In addition, the playback side must manage the time for searching and playing back such a variable-length data stream.

SUMMARY OF THE INVENTION It is therefore an object of the present invention to provide an audio encoding method capable of managing a processing time on a reproduction side when encoding a multi-channel audio signal at a variable compression rate.

[0007]

The present invention, in order to achieve the above object, comprises the following means. That is,

A multi-channel audio signal is obtained for each channel as it is or for each channel correlated with each other, in response to an input audio signal, a leading sample value is obtained,
Selecting a linear prediction method whose prediction residual is a minimum value among a plurality of prediction values of a current signal predicted from a past signal in the time domain and performing predictive encoding; and Generating decoding time stamp information indicating a timing of reading predicted encoded data (compressed data) in an input buffer on the decoding side according to a predicted encoded data amount for each channel; A voice encoding method comprising: a packet header including stamp information; and a step of formatting user data including the predicted encoded data into a packet.

[0009]

Embodiments of the present invention will be described below with reference to the drawings. FIG. 1 is a block diagram showing a first embodiment of a speech encoding apparatus to which the present invention is applied 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. FIG. 1 is an explanatory view showing a bit stream encoded by the encoding unit shown in FIG. 2, and FIG.
FIG. 5 is an explanatory diagram showing the format of a DVD audio pack, FIG. 6 is a block diagram showing the decoding unit of FIG. 1 in detail, and FIG. 7 is a write / read operation of the input buffer of FIG. FIG. 8 is a timing chart showing timing, 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 multi-channel systems. (1) Four-channel system As in the Dolby surround system, a total of four channels including three channels of front L, C, and R + one channel of rear S (2) Five-channel system S in the Dolby AC-3 system
Like without W channel, 3 channels of front L, C and R + 2 channels of rear SL and SR, total 5 channels (3) 6 channel system DTS (Digital Theater)
6) (L, C, R, SW (Lfe), SL, SR) like the Dolby AC-3 system (4) 8-channel system SDDS (Sony Dynamic D
digital sound), forward L, LC, C, R
6 channels of C, R, SW + 2 channels of rear SL, SR, total 8 channels

The 6-channel (ch) mixing and matrix circuit 1 'on the encoding side shown in FIG. 1 includes a front left (Lf), a center (C), a front right (Rf), a surround left ( Ls), surround light (Rs) and Lfe (Low Frequency Effe)
ct), the 6-channel PCM data is classified and converted into 2ch “1” and “2” for the front group and 4ch “3” to “6” for the other group by the following equation (1), and converted into 2ch “1”.
"2" is assigned to the first encoding unit 2'-1, and 4ch "3" to
"6" is output to the second encoding unit 2'-2.

As shown in detail in FIG. 2, the first and second encoding units 2'-1 and 2'-2 constituting the encoding unit 2 'respectively have 2ch "1", "2" and 4ch "3". "~" 6 "PC
The M data is predictively coded, and the predicted coded data is transmitted to the decoding side via the recording medium 5 and the communication medium 6 in a bit stream as shown in FIG. On the decoding side, the first and second decoding units 3'-1 and 3'-2 constituting the decoding unit 3 'respectively provide 2ch "1" and "2" for the forward group as shown in detail in FIG. 4ch for other groups
The predictive encoded data of “3” to “6” is decoded into PCM data.

Next, the original 6 ch (Lf, C, Rf, Ls,
Rs, Lfe) are restored, and stereo 2-ch data (L, R) is generated from the original 6 ch and coefficient mij (i = 1, 2, j = 1, 2 to 6) as in the following equation (2). I do. 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)

Referring to FIG. 2, encoding sections 2'-1, 2'-
2 will be described in detail. PC for each channel "1" to "6"
The M data is stored in one frame buffer 10 for each frame. Then, the sample data of each of the channels “1” to “6” of one frame are respectively supplied to the prediction circuits 13D1 and 13D
2, 15D1 to 15D4 and each channel
The first sample data of each frame of “1” to “6” is applied to the formatting circuit 19. Prediction circuit 13D
1, 13D2, 15D1 to 15D4 are each channel
For the PCM data of “1” to “6”, a plurality of linear prediction values of a current signal are calculated from a past signal in a time domain by a plurality of predictors (not shown) having different characteristics, and then the original PCM data Then, a prediction residual for each predictor is calculated from the plurality of linear prediction values. Following buffer / selector 14D
1, 14D2, 16D1 to 16D4 temporarily store the prediction residuals calculated by the prediction circuits 13D1, 13D2, 15D1 to 15D4, respectively, and select the 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 applies a bit number flag of the prediction residual to the packing circuit 18 and the formatting circuit 19, and outputs a predictor selection flag indicating a predictor having the minimum prediction residual. To the formatting circuit 19, the correlation coefficient a in the equation (1) and the DTS indicating the time at which the decoding side takes out the stream data from the input buffer 22a (FIG. 6). Packing circuit 18
Are buffer / selectors 14D1, 14D2, 16D1-1.
The prediction residual for 6 ch selected by 6D4 is packed with the specified bit number based on the bit number flag specified by the selection signal / DTS generator 17. In the PTS generator 17c, the decoding side is the output buffer 110 (FIG. 6).
A PTS (Presentation Time Stamp) indicating the time at which the PCM data is to be extracted from the PCM is generated and output to the formatting circuit 19.

The following formatting circuit 19 is shown in FIGS.
Format as user data as shown in FIG.
Is a variable-rate bit stream (sub-stream) BS0 including 2ch “1” and “2” prediction coded data related to the forward group.
And a variable-rate bit stream (substream) BS1 including 4ch “3” to “6” prediction coded data relating to other groups, and a bitstream header (restart header) provided before substreams BS0 and BS1 ). Also, substream B
One frame of S0 and BS1 includes: a frame header; first sample data of one frame of each channel “1” to “6”; and selection of a predictor for each subframe of each channel “1” to “6”. A flag; a bit number flag for each subframe of each ch “1” to “6”; a prediction residual data string (variable bit number) for each ch “1” to “6”; Are multiplexed. According to such predictive coding, when the original signal has, for example, a sampling frequency of 96 kHz, the number of quantization bits = 24 bits, and 6 channels, a compression ratio of 71% 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, FIG. The audio (A) pack shown is packed. This pack has 4 bytes of pack start information and 6 bytes of SCR (Syst) for 2034 bytes of user data (A packet, V packet).
em Clock Reference (system time reference value) information, a 3-byte Mux rate (rate) information, and a 1-byte stuffing that add a pack header of a total of 14 bytes (1 pack = 2048 bytes in total) . In this case, the time stamp SCR information is
In the first pack, the time of the A-pack in the same title can be managed by setting it to “1” so as to be continuous within the same title.

As shown in detail in FIG. 5, the A packet of the compressed PCM has a packet header of 19 or 14 bytes, a private header of the compressed PCM, and audio data (compressed PCM) of 1 to 2011 bytes in the format shown in FIG. It consists of. And DTS and P
The TS is set in the packet header of FIG. 5 (specifically, the PTS is set in the 10th to 14th bytes and the DTS is set in the 15th to 19th bytes). The private header of the compressed PCM is: 1-byte substream ID, 2 bytes of UPC / EAN-ISRC (Universal Prism).
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, 8-byte audio data information (ADI), and 0 to 7 stuffing bytes. ing. 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
Bytes (specifically, the forward access unit search pointer is set at the seventh byte of the ADI, and the backward access unit search pointer is set at the eighth byte).

Next, referring to FIG. 6, the decoding units 3'-1,
3′-2 will be described. The variable rate bit stream data BS0 and BS1 in the above format are separated by the deformatting circuit 21. And each channel
The head sample data of one frame of “1” to “6” and the predictor selection flag are stored in the prediction circuits 24D1 and 24D, respectively.
2, 23D1 to 23D4, and each channel “1” to
The bit number flag of “6” is 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, the prediction circuits 24D1, 2D2
A plurality of predictors (not shown) in 4D2 and 23D1 to 23D4 are respectively provided on the prediction circuits 13D1 and 13D on the encoding side.
2, the same characteristics as those of the plurality of predictors in 15D1 to 15D4, and those having the same characteristics are selected by the predictor selection flag.

The stream data (prediction residual data string) separated by the reformatting circuit 21 is fetched and stored in the input buffer 22a for each access unit by the SCR as shown in FIG. Here, the data amount of one access unit is, for example, fs = 96 kHz.
In the case of (1), it is (1/96 kHz) seconds, but as shown in detail in FIGS. 8 and 9A, the length is variable. The stream data accumulated in the input buffer 22a is DT
Based on S, the data is read out from the FIFO and applied to the unpacking circuit 22.

The unpacking circuit 22 is provided for each channel "1" to
The prediction residual data string of “6” is separated based on each bit number flag, and is divided into prediction circuits 24D1, 24D2, and 23, respectively.
It outputs to D1-23D4. Prediction circuits 24D1, 24D
2, 23D1 to 23D4, the current prediction residual data of each of the channels “1” to “6” from the unpacking circuit 22 and each of the plurality of internal predictors selected by the predictor selection flag. The previous predicted value predicted by one frame is added to calculate the current predicted value, and then the PC of each sample is determined based on the first sample data of one frame.
M data is calculated and stored in the output buffer 110. PCM data stored in the output buffer 110 is P
Read and output based on TS. Therefore,
The variable length access unit shown in FIG. 9A is decompressed, and a fixed length presentation unit shown in FIG. 9B is output.

Here, when a search reproduction is instructed via the operation unit 101, the control unit 100 sets a forward access unit search pointer indicating one second ahead in the ADI shown in FIG. The access unit is reproduced on the basis of the backward access unit search pointer indicating. 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 the processing shown in FIG. (Step S41), add a packet header (step S42), and send this packet out onto the network (step S43).

On the decoding side, the header is removed as shown in FIG. 11A (step S51), the data is restored (step S52), and the data is stored in a memory and decoding is waited (step S53). . Then, when decoding is performed, as shown in FIG. 11B, deformatting is performed (step S61), and then the input buffer 22
The input / output control of a is performed (step S62), and then the 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), input / output control of the output buffer 110 is performed (step S65), and the original multi-channel is restored (step S66). This is output (step S67), and thereafter, this is repeated.

In the above embodiment, 2ch "1" and "2" relating to the front group are converted by "1" = Lf + Rf "2" = Lf-Rf and are predictively coded. Multi-channel is downmixed to generate stereo 2ch data (L, R), and then the following equation (1) ′ And predictive encoding may be performed (second
Embodiment). In this case, the mix & matrix circuit 4 ′ on the decoding side adds the channels “1” and “2” to the channel L, and subtracts the channel R by adding the channels “1” and “2”.
Can be generated.

As a third embodiment, as shown in FIG. 12, multi-channel downmixing is performed according to equation (2) instead of 2ch "1" and "2", and stereo 2ch data (L, R) is obtained. The stereo 2ch (L, R) and the 4ch “3” to “6” may be generated and 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 encoding and select predictive encoded data of a group having the highest compression ratio. Therefore, the encoding unit shown in FIG. 13 includes first to n-th correlation circuits 1-1 to 1-n,
The n correlation circuits 1-1 to 1-n have, for example, 6 channels (L
f, C, Rf, Ls, Rs, and Lfe) PCM data are converted into n types of 6-channel 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 Further, 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 the group having the highest compression ratio is determined based on the data amount of the minimum prediction residual for each group. Are selected by the correlation selection 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).

On the decoding side shown in FIG. 14, n correlating circuits 4 are provided for the correlating circuits 1-1 to 1-n on the encoding side.
−1 to 4-n (or one correlation circuit (not shown) whose coefficients a and b can be changed) are provided. Note that n shown in FIG.
When the prediction circuits of the groups have the same configuration, the decoding device does not need to provide the prediction circuits of n groups as shown in FIG. 14, and may use the prediction circuits of one group. 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 selected.
And set the original 6 ch (Lf, C, Rf, Ls, Rs, L
fe), and down-mixes the multi-channels according to equation (2) to generate stereo 2-ch data (L, R).

In the first embodiment, one kind of correlation signal "1" to "6" is configured to be predictively coded. However, the signals "1" to "6" are encoded. And the group of the original signals (Lf, C, Rf, Ls, Rs, Lfe) may be predictively coded and the group with the higher compression ratio may be selected.

[0032]

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

[Brief description of the drawings]

FIG. 1 is a block diagram showing 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 showing a bit stream encoded by an encoding unit shown in FIGS. 1 and 2;

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.

FIG. 6 is a block diagram illustrating a decoding unit of FIG. 1 in detail;

FIG. 7 is a timing chart showing write / read timings of the input buffer of FIG. 6;

FIG. 8 is an explanatory diagram showing the amount of compressed data for each access unit.

FIG. 9 is an explanatory diagram showing an access unit and a presentation unit.

FIG. 10 is a flowchart showing a voice transmission method.

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

FIG. 12 is a block diagram showing a third embodiment of a speech coding apparatus to which the present invention is applied and a speech decoding apparatus corresponding thereto.

FIG. 13 is a block diagram illustrating a speech encoding device according to a fourth embodiment.

FIG. 14 is a block diagram illustrating a speech decoding device according to a fourth embodiment.

[Explanation of symbols]

1 '6ch Mix & Matrix Circuit 13D1, 13D2, 15D1-15D4 Prediction Circuit (Buffer / Selector 14D1, 14D2, 16D1-1
A compression means is constituted together with 6D4. 14D1, 14D2, 16D1-16D4 buffer
Selector 17 Selection signal / DTS generator (timing generating means) 17c PTS generator (timing generating means) 19 Formatting circuit (Formatting means) 21 Deformatting circuit (Separating means) 22 Unpacking circuit 22a Input buffer 24D1, 24D2, 23D1 to 23D4 Prediction circuit (expansion means) 100 Control unit (reading means) 110 Output buffer

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[Procedure amendment]

[Submission date] February 28, 2001 (2001.2.2
8)

[Procedure amendment 1]

[Document name to be amended] Statement

[Correction target item name] Claims

[Correction method] Change

[Correction contents]

[Claims]

[Procedure amendment 2]

[Document name to be amended] Statement

[Correction target item name] 0008

[Correction method] Change

[Correction contents]

In response to an input audio signal, a multi-channel audio signal is obtained for each channel as it is or for each channel correlated with each other , a leading sample value is obtained, and time is calculated by a plurality of linear prediction methods having different characteristics. Predict linear predictions of the current signal from past signals in the region
From the predicted linear prediction value and the audio signal
Selecting and compressing a linear prediction method that minimizes the obtained prediction residual, and timing for reading compressed data in the input buffer on the decoding side according to the amount of the compressed data. generating a decoding timestamp information indicating a packet header containing the decoding timestamp information, comprising the steps of formatting user data into packets comprising a said compressed <br/> data , A speech coding method.

Claims (1)

[Claims] A multi-channel audio signal is obtained for each channel as it is or for each channel correlated with each other by obtaining a leading sample value in response to an input audio signal.
A step of selecting a linear prediction method whose prediction residual is a minimum value among a plurality of prediction values of a current signal predicted from a past signal in the time domain and performing predictive encoding; and Generating decoding time stamp information indicating timing for reading predicted encoded data (compressed data) in an input buffer on the decoding side according to a predicted encoded data amount for each channel; A voice encoding method comprising: a packet header including stamp information; and a step of formatting user data including the predicted encoded data into a packet.