JPH09102742A - Encoding method and device, decoding method and device and recording medium - Google Patents

Abstract

PROBLEM TO BE SOLVED: To encode the multichannel signals with high efficiency. SOLUTION: A mixer 102 produces the mixture processing data based on the audio data supplied through the input terminals 101a to 101e, and these processing data are supplied to the processing data extractors 103a to 103e and the encoders 105f and 105g respectively. The extractors 103a to 103e supply the data which are not reproduced by the mixture processing data to the encoders 105a to 105e, and the reproduction parameters of other data which are reproduced by the mixture processing data are supplied to a multiplexer 106. The reproduction parameters and the data which are encoded by the encoders 105a to 105g are turned into a bit stream by the multiplexer 106 and then outputted.

Description

DETAILED DESCRIPTION OF THE INVENTION

[0001]

TECHNICAL FIELD The present invention relates to an encoding method and apparatus, a decoding method and apparatus, and a recording medium,
For example, a stereo such as a video tape recorder, a video disc player, or a so-called multi-sound acoustic system, which is suitable for use in compressing and encoding a multi-channel digital signal, an encoding method and apparatus, a decoding method and apparatus, and Recording medium

[0002]

2. Description of the Related Art Conventionally, there are various techniques and apparatuses for highly efficient encoding of signals such as audio or voice.
For example, an audio signal in the time domain is divided into blocks for each unit time, the time axis signal of each block is converted into a signal on the frequency axis (orthogonal conversion), and divided into a plurality of frequency bands. Blocking frequency band division method for encoding, so-called transform coding (transform coding), or non-blocking for dividing into a plurality of frequency bands and encoding without dividing the time domain audio signal into blocks for each unit time Band division coding (sub-band coding: SBC), which is a generalized frequency band division method, can be used.

Further, a high-efficiency coding method and apparatus combining the above-described band-division coding and transform coding has been considered. In this case, for example, band-division is performed by band-division coding. After that, the divided signal for each band is orthogonally transformed into a signal in the frequency domain, and the orthogonally transformed signal is encoded for each region.

Here, as the band division filter used in the above band division encoding, for example, QMF is used.
There is a filter such as (Quadrature Mirror Filter), and this QMF filter is referred to as "Digital coding of speech in subbands" R.
E. Crochiere, BellSyst.Tech. J., Vol.55, No.8 1976)
It is described in. This QMF filter divides the band into two equal bandwidths, and this filter is characterized in that so-called aliasing does not occur when the divided bands are combined later.

In addition, the document "Polyphase Quadrature Filters-New Band Division Coding Technique"("Poly
phase Quadrature filters -A newsubband coding tech
nique ", oseph H. Rothweiler ICASSP 83, BOSTON) describes a method for dividing equal-bandwidth filters. In this polyphase quadrature filter, the signal is divided into multiple bands of equal bandwidth. The feature is that it can be divided at one time.

Further, as the spectrum transformation of the above-mentioned orthogonal transformation, for example, an input audio signal is divided into blocks in a predetermined unit time (frame), and discrete Fourier transform (DFT), discrete cosine transform (DCT), or modified for each block. There is one that transforms a time axis into a frequency axis by performing discrete cosine transform (MDCT) or the like. Regarding the MDCT, reference is made to the document "Subband / Transform Coding Using Filter Bank Design Based on Time Domain Aliasing Cancellation"("Subband / Transform
Coding Using Filter Bank Designs Based on Time Dom
ain Aliasing Cancellation ", JPPrincen ABBradla
y, Univ.of Surrey Royal Melbourne Inst.of Tech. IC
ASSP 1987).

In this way, by quantizing the signal divided for each band by the filter and the spectrum conversion,
It is possible to control the band in which the quantization noise is generated, and it is possible to perform auditory and more efficient encoding by utilizing the properties such as the so-called masking effect. Further, if the normalization is performed for each band, for example, with the maximum absolute value of the signal component in that band before the quantization is performed here, more efficient encoding can be performed.

Here, as the frequency division width in the case of quantizing each frequency component divided into frequency bands, for example, a bandwidth considering human auditory characteristics is often used. That is, the audio signal may be divided into a plurality of bands (for example, 25 bands) with a bandwidth generally called a critical band in which the higher the band, the wider the bandwidth. Further, at the time of encoding the data for each band at this time, encoding is performed by predetermined bit allocation for each band or adaptive bit allocation (bit allocation) for each band. For example, the MDC
When the coefficient data obtained by the T processing is encoded by the bit allocation, the number of adaptive allocation bits (adaptation to the MDCT coefficient data for each band obtained by the MDCT processing for each block) The encoding is performed with the number of distributed bits).

The following two methods are known as the bit allocation method (bit allocation method).

For example, the document "Adaptive Transform Coding of Speech Signals",
IEEE Transactions of Acoustics, Speech, and Signa
Processing, vol.ASSP-25, No.4, August 1977), bit allocation is performed based on the signal size of each band.

In addition, for example, the document "Critical band encoder-
Digital Encoding of Perceptual Requirements of the Auditory System "(" The critical band coder --digital encodi
ng of the perceptual requirements of the auditory
system ", MA Kransner MIT, ICASSP 1980) describes a method that uses auditory masking to obtain a necessary signal-to-noise ratio for each band and perform fixed bit allocation.

[0012]

By the way, for example, in the high-efficiency compression encoding system for audio signals using the above-mentioned sub-band coding or the like, the human auditory characteristic is utilized to convert the audio data into about 1 A method of compressing to / 5 has already been put into practical use. As a high-efficiency encoding method for compressing the audio data to about 1/5, for example, ATRAC (a trademark of Sony Corporation, Adaptive, which is used in MD (a trademark of Sony Corporation, Mini Disc) standard).
There is a method called TRansform Acoustic Coding).

Not only in the case of ordinary audio equipment but also in stereo or multi-sound sound systems such as movie film projection systems, high-definition televisions, video tape recorders, video disc players, etc., for example, 4 to 8 channels, etc. In this case, audio signals or voice signals of a plurality of channels are being handled, and even in this case, it is desired to perform high efficiency coding that reduces the bit rate.

In addition to this, in addition to audio signals or audio signals of a plurality of channels, in order to enable reproduction even in an existing stereo or acoustic system, audio signals of a plurality of channels such as 4 to 8 channels or In some cases, the audio signal may be converted into, for example, 2-channel data by a method such as down-mixing, and the converted 2-channel data may be recorded separately from the audio signals of the plurality of channels.

FIG. 10 shows an example of the configuration of a multi-channel coding apparatus for compressing and coding the signals of the respective channels, and simultaneously coding the signals obtained by subjecting the signals of the respective channels to processing such as mixing. Is shown.

The encoder shown in FIG. 10 has an input terminal 1
Mixers 102 for mixing the signals of the respective channels input from 01a to 101e and converting them into signals of the two channels, encoders 105a to 105e for encoding the signals of the respective channels, and mixers supplied by the mixer 102. Encoders 105f and 105g for encoding the generated signal, encoder 10
It comprises a multiplexer 106 for converting the coded signals from 5a to 105g into a bit stream, and an output terminal 107 for outputting the bit stream from the multiplexer 106.

For example, a center (C) channel, a left (L) channel, a right (R) channel, a left surround (SL) channel, and a right surround (S) supplied through the input terminals 101a to 101e.
Each audio data of the R) channel is supplied to each of the single-channel encoders 105a to 105e. In these encoders 105a to 105e, the input signal is encoded, and the encoded data is supplied to the multiplexer 106. The multiplexer 106 converts the encoded data of each channel into one bit stream, and this bit stream is output to the output terminal 107.
Output from

On the other hand, the audio data input from the input terminals 101a to 101e is also supplied to the mixer 102, and, for example, the mixing processing is performed at the following ratio,
Reconstructed as channel data.

That is, in one channel (L _mix channel), the L channel, the R channel, the C channel, the SL channel, and the SR channel are 1.0000 and 0.0.
000, 0.7071, 0.7071, and 0.00
It is mixed in the ratio of 00. The other channel (R _mix channel) is L channel, R channel, C channel, SL
Channels and SR channels are 0.0000, 1.
0000, 0.7071, 0.0000, and 0.7
071 mixed.

The two-channel data reconstructed by these mixing processes are supplied to the encoders 105f and 105g and encoded. Encoders 105f and 105g
The encoded data in 1 is supplied to the multiplexer 106, and the encoded data of the above two channels is converted into one bit stream, and then output from the output terminal 107.

FIG. 11 shows an example of the configuration of a multi-channel decoding device that performs decoding for each channel. Further, FIG. 12 shows a configuration example of a two-channel decoding device that decodes a signal in which a part or all of digital signals of a plurality of channels are mixed.

The decoding device shown in FIG. 11 has an input terminal 1
Demultiplexer 132 that divides the bit stream input from 31 into encoded data of each channel, decoders 133a to 133e that respectively decode encoded data corresponding to each channel from demultiplexer 132,
And output terminals 136a to 136e for outputting the signals decoded by the decoders 133a to 133e.

The encoded bit stream data supplied through the input terminal 131 is the demultiplexer 1
At 32, the data is divided into encoded data corresponding to each channel and is supplied to each of the decoders 133a to 133e. The encoded data supplied to the decoders 133a to 133e are decoded in the decoders 133a to 133e, respectively, and the decoded audio data are output from the output terminals 136a to 136e, respectively.

In the decoding apparatus shown in FIG. 12, the demultiplexer 132 for dividing the encoded bit stream data input from the input terminal 131 into the mixed two-channel encoded data is subjected to the mixing processing. Decoder 13 for decoding each of the two-channel encoded data
It is composed of 3f and 133g.

The encoded bit stream data supplied through the input terminal 131 is the demultiplexer 1
In 32, the mixed data is divided into two-channel encoded data. The divided 2-channel encoded data is supplied to the decoders 133f and 133g, respectively.
The encoded data supplied to the decoders 133f and 133g are decoded, and the decoded audio data is output from the output terminals 136f and 136g, respectively.

As described above, in the case of recording, for example, a two-channel signal in which these signals are mixed and processed separately from the audio signal or the voice signal of a plurality of channels, a high efficiency code for further reducing the bit rate. It is desired to implement

However, in the encoding device and the decoding device having the configurations shown in FIGS. 10, 11 and 12, the digital audio data is reduced to about 1 /
The high-efficiency coding method of compressing to 5 is a coding method for a single channel, and when coding multi-channel audio data using this, data dependence between channels and data of each channel It is not possible to perform effective data coding processing using elements such as characteristics and format characteristics. That is, it is impossible to use an element that the audio data of the C channel and the audio data of the L channel are similar to each other, or the audio data of the left and right surround channels are similar in terms of format.

The present invention has been made in view of such a situation, and in the compression coding of multi-channel signals, high compression suitable for the degree of correlation of digital data between multi-channels has been achieved. It can be realized by using an encoder and a decoder.

[0029]

According to a first aspect of the present invention, there is provided at least one frequency band of at least one part of a digital signal of at least one channel corresponding to a characteristic of the digital signal and a reproduction environment. Individual mixed coded digital signals that are mixed into two mixed channels and are not encoded by the mixed digital signals of the mixed channels are reproduced. It is characterized by

According to a second aspect of the present invention, the encoding device has at least 1 in accordance with the characteristics of the digital signal and the reproduction environment.
Mixing means for mixing part or all of the frequency bands of the digital signals of one channel into at least one mixing channel, and individually encoding the digital signal excluding the signal reproduced by the mixed digital signal of the mixing channel It is characterized in that it comprises extraction means for extracting the encoded digital signal and encoding means for encoding the mixed digital signal and the individual encoded digital signal.

The decoding method according to claim 11 is for decoding the mixed digital signal based on the coded information and for restoring the digital signal by using a part or all of the decoded mixed digital signal. It is characterized in that the data for restoration is created, the individually encoded digital signal and the data for restoration are combined, and the digital signal is restored.

According to a twelfth aspect of the present invention, there is provided a decoding device for decoding the mixed digital signal based on the coded information, and a part or all of the mixed digital signal decoded by the decoding means. Using the restoring data creating means for creating the restoring data for restoring the digital signal, the restoring data created by the restoring data creating means, and the individual encoded digital signal are combined to restore the digital signal. And a restoring means for performing the restoration.

In the recording medium according to the eighteenth aspect, a part or all of the frequency bands of the digital signals of at least one channel of the plurality of channels are mixed in at least one mixing channel, and the frequency characteristics of the digital signal and the reproduction environment. Corresponding to, the individual coded digital signal to be individually coded is extracted from the digital signal excluding the signal reproduced by the mixed digital signal of the mixed channel, and is mixed based on the predetermined coded information. A digital signal and an individually encoded digital signal are recorded.

In the encoding method according to claim 1,
Depending on the characteristics of the digital signal and the reproduction environment, some or all of the frequency bands of the digital signal of the at least one channel are mixed into the at least one mixed channel and reproduced from the digital signal by the mixed digital signal of the mixed channel. The individual coded digital signals to be coded separately from which the signals have been removed are extracted, and the mixed digital signal and the individual coded digital signals are coded. Therefore, the original digital signal can be reproduced based on the mixed digital signal, and the compression rate of the digital signal can be increased.

In the encoding device according to the second aspect,
Depending on the characteristics of the digital signal and the reproduction environment, some or all of the frequency bands of the digital signal of the at least one channel are mixed by the mixing means into the at least one mixing channel, and the mixing means mixes the digital signal by the extraction means. The individual coded digital signal to be individually coded is extracted by removing the signal reproduced by the mixed digital signal of the channel, and the mixed digital signal and the individual coded digital signal are coded by the coding means. Therefore, the original digital signal can be reproduced based on the mixed digital signal, and the compression rate of the digital signal can be increased.

In the decoding method according to the eleventh aspect, the mixed digital signal is decoded based on the coded information, and the digital signal is restored by using a part or all of the decoded mixed digital signal. Restoration data is created, and the individually encoded digital signal and the restoration data are combined to restore the digital signal. Therefore, the original digital signal can be restored based on the mixed digital signal.

In the decoding device according to the twelfth aspect of the present invention, the decoding means decodes the mixed digital signal based on the coded information, and the restoration data creating means makes part or all of the mixed digital signal. Is used to create restoration data for restoring the digital signal, and the restoration means synthesizes the restoration data and the individual encoded digital signal to restore the digital signal. Therefore, the original digital signal can be restored based on the mixed digital signal.

In the recording medium according to claim 18,
The mixed digital signal and the individually encoded digital signal encoded based on predetermined encoded information are recorded.
Therefore, a high efficiency coded digital signal is recorded,
You can play it.

[0039]

BEST MODE FOR CARRYING OUT THE INVENTION Embodiments of the present invention will be described below with reference to the drawings.

FIG. 1 is a block diagram showing the configuration of an embodiment of an encoding device to which the encoding method of the present invention is applied.
As shown in the figure, the encoding apparatus of the present invention is a single-channel compression encoder (for example, the so-called A described above).
A plurality of TRAC encoders are used to realize multi-channel compression encoding.

That is, the encoding device shown in FIG. 1 is a device for encoding digital audio signals of a plurality of channels and outputting encoding parameter information together with the encoded digital audio signals. Means) mixes some or all of the digital signals of the plurality of channels. Processed data extractors 103a to 10
3e (extracting means) compares the mixed digital signal (mixing processing data) supplied from the mixer 102 with the audio data supplied from the input terminals 101a to 101e, and from this audio data, the mixing processing data is obtained. Separately, audio data (individually encoded data) determined to be more suitable for individual encoding is extracted and supplied to the encoders 105a to 105e (encoding means).
Further, for the remaining audio data for which it is judged that it is more effective to reproduce the original audio data by the mixed processing data supplied from the mixer 102, a reproduction parameter for reproducing the remaining audio data is generated, It is adapted to be supplied to the multiplexer 106.

The encoders 105a to 105e encode the individual encoded data supplied from the processed data extractors 103a to 103e and output the encoded data. The encoders 105f and 105g are configured to encode and output the mixed processing data supplied from the mixer 102.

The multiplexer 106 has an encoder 105a.
To 105g, the encoded data and the reproduction parameter supplied from the first to 105 g are converted into one bit stream, which is then output from the output terminal 107.

Here, the center (C) channel, the left (L) channel, the right (R) channel, the left surround (SL) channel, and the right surround (S).
The description will be made assuming that the R channel 5-channel audio data is input from the input terminals 101a to 101e, respectively. Further, the mixer 102 performs a mixing process on these 5-channel audio data to create 2-channel audio data, and outputs the 2-channel audio data.

In the embodiment shown in FIG. 1, the center (C), the left (L), the right (R), the left surround (SL), and the right surround (SR) supplied via the input terminals 101a to 101e. The audio data of each channel is first input to the mixer 102. The mixer 102 is basically the same as the mixer 102 shown in FIG. 10, and a detailed description thereof will be omitted. However, as described above with reference to FIG.
The audio data of each channel input from 01a to 101e are mixed and reconfigured as two-channel data. The output of the mixer 102 is supplied to the encoders 105f to 105g and the processed data extractor 10
It is also supplied to 3a to 103e.

In the processed data extractors 103a to 103e, the output from the mixer 102 is compared with the audio data supplied from the input terminals 101a to 101e for each channel, and it is judged appropriate to individually encode. The audio data thus extracted is extracted and output to the encoders 105a to 105e. On the other hand, for the remaining audio data for which it is determined that it is more effective to reproduce the original audio data by the output from the mixer 102, a reproduction parameter is generated and supplied to the multiplexer 106. Detailed configurations and operations of the processed data extractors 103a to 103e will be described later.

In the encoders 105a to 105e,
The audio data (individually encoded data) to be encoded individually supplied from the processed data extractors 103a to 103e are each encoded and supplied to the multiplexer 106. In the encoders 105f to 105g, the output data (mixing processing data) from the mixer 102 is encoded and supplied to the multiplexer 106.
The encoders 105a to 105g are basically the same as the encoders 102a to 102g shown in FIG. The detailed configuration and operation will be described later.

In the multiplexer 106, the encoded data supplied from the respective encoders 105a to 105g and the processed data extractors 103a to 103e.
The parameters for reproduction supplied by the above are converted into one bit stream and output from the output terminal 107.

FIG. 2 shows an internal configuration example of the processed data extractor 103a in the embodiment of FIG. The configurations and operations of the processed data extractors 103b to 103e are basically the same as those of the processed data extractor 103a, and therefore description thereof will be omitted.

The band division filters 202a and 202b constituting the processed data extractor 103a are connected to the input terminal 201.
The center channel audio data input from a is divided for each frequency band in which it is analyzed. The processed data analyzers 204a and 204b are provided for each frequency band that uses the mixed processed data as a reproduction signal for reproducing the original audio data,
The audio data of the channel is compared with the mixed processing data, and for the frequency band restored from the mixed processing data, an effective parameter (for example, a scale factor) for restoring the audio data of the channel is analyzed,
The analysis result is output to the processed data generator 205.

The parameter recording memory 203a stores the parameter as the analysis result supplied from the processed data analyzer 204a. Similarly, the parameter recording memory 2
03b stores parameters as analysis results supplied from the processed data analyzer 204b.

Based on the outputs from the processed data analyzers 204a and 204b, the processed data generator 205 extracts only the frequency band component required for encoding from the audio data of the channel input from the input terminal 201a, Output from the output terminal 206a. Further, the parameters for restoring the audio data of the channel, which are supplied from the processed data analyzers 204a and 204b, are put together into one and output from the output terminal 206b.

The center channel audio data from the input terminal 101a in FIG. 1 is input to the input terminal 201a, and the mixer 10 is input to the input terminals 201b and 201c.
When the mixed processing data from 2 is input, the input terminal 20
The data input via 1a, 201b, and 201c are respectively supplied to band division filters 202a to 202b, divided into frequency bands (for example, critical bands) for analyzing the audio data, and processed data analyzers 204a to 204a. It is supplied to 204b.

Processed data analyzers 204a-204b
Is for each frequency band that uses the mixed processed data as a reproduction signal for reproducing the original audio data, compares the audio data of the channel and the mixed processed data, and mixes them. A parameter (for example, a scale factor) effective for restoring the audio data of the channel from the processed data is analyzed, and the analysis result is output to the processed data generator 205.

Since the analysis result of the previous frame is used for the analysis of this parameter, the analysis result is also supplied to and recorded in the parameter recording memories 203a and 203b for storing the analysis result. In the processed data analyzers 204a and 204b, the analysis results recorded in the parameter recording memories 203a and 203b are appropriately used as needed to analyze the parameters.

In the processed data generator 205, based on the outputs from the processed data analyzers 204a and 204b, only the frequency band component necessary for encoding is extracted from the audio data of the channel input from the input terminal 201a. And is output from the output terminal 206a. Further, the parameters for restoring the audio data of the channel, which are supplied from the processed data analyzers 204a and 204b, are put together into one and output from the output terminal 206b.

Therefore, with reference to the parameters of the previous frame, the mixed-processed channel to be used as the reproduction signal, or the ratio of its use, can be changed on a frame-by-frame basis, or the frequency band for coding the data of the channel can be changed. It is possible to change or change the value of an appropriate parameter for each frequency band, and it is possible to reduce the discomfort when restored.

FIG. 3 is a block diagram showing the configuration of an embodiment of a decoding device to which the decoding method of the present invention is applied.
The decoding apparatus shown in FIG. 3 realizes multi-channel decoding by using a plurality of single-channel compression code decoders (for example, decoders corresponding to the so-called ATRAC system described above).

The demultiplexer 132 has the input terminal 13
The encoded audio data of each channel included in the bit stream input from 1 and the reproduction parameter for reproducing the original audio data from the mixed processed data are divided for each corresponding channel. Then, the encoded audio data is supplied to the decoders 133a to 133e (decoding means) of the corresponding channels, and the reproduction parameters are combined data generators 134a to 134e (restoring data generating means) of the corresponding channels. ) Respectively. Further, the encoded mixed processing data is supplied to the decoders 133f and 133g.

The decoders 133a to 133e decode the encoded audio data supplied from the demultiplexer 132, and supply the decoded audio data to the synthesizers 135a to 135e (restoring means). The decoders 133f and 133g are configured to decode the mixed processing data supplied from the demultiplexer 132 and supply it to the corresponding combined data generators 134a to 134e, respectively.

Combined data generators 134a to 134e
Is based on the reproduction parameters supplied from the demultiplexer 132 from the mixed processing data supplied from the decoders 133f and 133g, and creates reproduction data for reproducing the audio data of the relevant channel. It is adapted to be supplied to each of 135a to 135e.

The synthesizers 135a to 135e synthesize the reproduction data supplied from the synthesized data generators 134a to 134e and the decoded audio data supplied from the demultiplexer 132, and generate the original audio data of each channel. And output terminals 136a through 136
It is designed to be output from e.

From the input terminal 131 to the demultiplexer 13
When the bitstream is input to the demultiplexer 1, the demultiplexer 1 receives the audio data of each channel and the parameters for reproducing the original data from the mixed processing data.
In 32, the encoded audio data, the mixed processing data subjected to the mixing processing, and the reproduction parameter are divided for each channel, and the encoded audio data and the mixed processing data subjected to the mixing processing are decoded by the decoder 13
3a to 133g, respectively, and the reproduction parameters are supplied to the combiners 135a to 135e, respectively.

Decoder 1 corresponding to the mixed-processed channel
In 33f and 133g, the encoded mixed processing data is decoded, and combined data generators 134a to 134
e respectively. In the synthesized data generators 134a to 134e, the original data is reproduced from the decoded mixed processing data supplied from the decoders 133f and 133g and the mixed processing data supplied from the demultiplexer 132. Data for reproducing the channel is created using the parameters, and the synthesizer 135a
To 135e. Synthetic data generator 134a
The detailed configurations and operations of the to 134e will be described later.

In the synthesizers 135a to 135e,
The audio data of each channel decoded by the decoders 133a to 133e and the data for reproducing each channel created by the combined data creating units 134a to 134e are respectively combined to obtain the decoded audio data of each channel. The output terminals 136a to 13a are respectively restored.
It is output from 6e.

FIG. 4 shows the combined data generator 134a of FIG.
2 shows an example of the internal configuration of FIG. Composite data generator 134b
To 134e are the same as those of the synthetic data generator 1
Since it is basically similar to the case of 34a, its description is omitted.

The band division filters 212a and 212b constituting the combined data generator 134a are connected to the input terminal 211.
The mixed processing data of the two channels input from a and 211b is divided for each frequency band in which reproduction data is created and output. The parameter analyzer 213 analyzes the reproduction parameter input from the input terminal 211c, divides it for each frequency band, and outputs it.

Reproduction data generators 214a and 214b
Converts the decoded mixed processing data for each frequency band based on the reproduction parameter and outputs the converted mixed processing data. The reproduction data synthesizer 215 synthesizes the outputs from the reproduction data generators 214a to 214e for each frequency band, and outputs the output terminal 2
It is designed to output from 16.

Decoded mixed processed data from the decoders 133f and 133g shown in FIG. 3 are input to the input terminals 211a and 211b, and the input mixed data from the demultiplexer 132 shown in FIG. 3 is input to the input terminal 211c. Channel reproduction parameters are input. Input terminals 211a, 21
The decoded mixed processing data supplied via 1b are supplied to the band division filters 212a to 212b, respectively, and are divided into frequency bands (for example, critical bands) for creating reproduction data, and then reproduction data creation is performed. To the containers 214a and 214b, respectively.

On the other hand, the parameter for reproduction of the channel input from the input terminal 211c is the parameter analyzer 2.
13, the parameters are analyzed and divided into frequency bands for analysis, and a reproduction data generator 214 for the corresponding band is generated.
a to 214b, respectively.

In the reproduction data generators 214a and 214b, the decoded mixed processing data is converted for each frequency band based on the reproduction parameters supplied from the parameter analyzer 213, and the reproduction data synthesizer 215 is used. Is supplied to.

In the reproduction data synthesizer 215, the outputs from the reproduction data generators 214a to 214b for each frequency band are synthesized, the reproduction data of the corresponding channel is generated, and then output from the output terminal 216.

As described above, in the above embodiment, the combined data generators 134a to 134e convert the mixed processing data into reproduction data in accordance with the optimum reproduction method determined by the encoding device, and the combiner data is reproduced. The original audio data is restored by combining with the data of the channel by 135. Therefore, by using the mixed processing data obtained by the mixing processing, it is possible to reduce the bit rate of the channel.

Next, another embodiment of the present invention will be described. In the above-described embodiment, the case where the number of the mixed-processed channels is two is described, but the present invention can be used as long as at least one channel is the mixed-processed channel.

The channel selection of the above embodiment is
The case of 5 channels of the center (C), the left (L), the right (R), the left surround (SL), and the right surround (SR) is described, and the left center (LC) channel and the right center ( For example, in the case of 7 channels including the RC) channel, in the case of 4 channels without the center (C) channel, and in the case where a subwoofer (SW) channel is further added, digital signals of a plurality of channels are encoded, In addition to this, it can be used when encoding a multi-channel digital signal having a channel obtained by mixing these.

Next, the above-mentioned encoders 105a to 105
The specific configuration and operation of e will be described with reference to FIGS. Note that FIG. 5 shows a configuration example of the encoder 105a for one channel. The encoder 105
Since the configurations and operations of b to 105e are basically the same as those of the encoder 105a, the description thereof is omitted here.

The signal band division filter 401 is adapted to divide the signal inputted through the input terminal 424 into three frequency bands. Low frequency MDCT (Modified D
The iscrete Cosine Transform (improved discrete cosine transform) circuit 402L is 0 supplied from the band division filter 401.
MDCT for low frequency signals from kHz to 5.5 kHz
Perform the operation. The mid-range MDCT circuit 402M has 5.5 kHz to 11 kHz supplied from the band division filter 401.
MDCT operation is performed on the signal in the middle band of z. High range MD
The CT circuit 402H is configured to perform MDCT calculation on a high frequency signal of 11 kHz or higher (11 kHz to 22 kHz) supplied from the band division filter 401.

The block size evaluator 403 determines a time block length described later. Normalization circuits 404L to 4
04H divides the audio signal consisting of low, mid, and high frequencies into a total of 52 block floating units of low, mid, and high frequencies, and normalizes (normalizes) each unit. Has been done.

The bit allocator 405 obtains allocation bit number information, which will be described later, and supplies it to the requantizer 406. The requantizer 406 is adapted to perform requantization based on the distributed bit number information supplied from the bit distributor 405. The formatter 407 is adapted to convert the output signal from the requantizer 406 into a bitstream.

In FIG. 5, the input terminal 424 is supplied with audio data (sampled and quantized audio data). The data supplied to the input terminal 424 is first supplied to the band splitting filter 401 through 0 to 5.
Low frequency of 5kHz, medium frequency of 5.5kHz to 11kHz, 11kHz or more (11kHz to 22kHz) 3
It is divided into two frequency bands.

Of these three frequency band signals, the low frequency signal from the band division filter 401 is input to the MDCT circuit 402L which performs MDCT operation, and the middle frequency signal is also M.
The MDCT circuit 402M that performs the DCT operation and the high-frequency signal are supplied to the MDCT circuit 402H.
Each of the CT circuits 402L to 402H is decomposed into frequency components. At this time, the time block length when performing MDCT processing is variable for each frequency band,
In the part where the signal changes abruptly, the time block length is shortened to improve the time resolution, and in the part where the signal is stationary, the time block length is lengthened to control the effective transmission of the signal component and the quantization noise. I have to.

The time block length is determined by the block size evaluator 403. That is, the signals of the three frequency bands from the band division filter 401 are also supplied to the block size evaluator 403, the block size evaluator 403 determines the time block length of MDCT processing, and indicates the determined time block length. Information is MDCT circuit 402
L to 402H are supplied.

Of the two types of time block lengths in MDCT processing, the long time block length is called the long mode and corresponds to a time of 11.6 ms, for example. The short time block length is called a short mode. For example, up to 1.45 ms in the high range (11 kHz or more), low range (5.5 kHz or less) and middle range (5.5 kHz to 1).
At 1 kHz), the time resolution is increased up to 2.9 ms.

In this way, the two-dimensional area of time and frequency (this is a block floating unit: Block Fl
The audio signal decomposed into the above signal components is divided into a total of 52 block floating units in the low band, the middle band, and the high band by the normalization circuits 404L to 404H, and standardized for each unit. Is normalized (normalized) (scale factor is determined).

Further, in the bit allocator 405, the characteristics of the audio signal are analyzed by utilizing the characteristics of human hearing. The analysis result is supplied to the requantizer 406 to which the signals for each unit from the normalization circuits 404L to 404H are supplied.

In the requantizer 406, the degree of accuracy with which each unit is coded is determined based on the above analysis result, parameterized (word length is determined), and re-quantized. Quantization is performed.

Finally, in the formatter 407,
The parameter information for each unit and the requantized spectrum signal are assembled into a bit stream in one channel supplied to the multiplexer 106 shown in FIG. 1 according to a predetermined format. The output of the formatter 407 is output from the output terminal 425.

Here, the above-described encoding operation is performed for each unit called a sound frame.

Further, the bit allocator 405 has a concrete structure as shown in FIG. That is, the energy calculation circuit 522 is configured to calculate energy for each critical band of the signal input from the input terminal 521. Convolution filter circuit 523
Performs a convolution process of multiplying an output signal from the energy calculation circuit 522 by a predetermined weighting function and adding the product.

The (n-ai) function generating circuit 525 generates and outputs an allowable function (n-ai). Subtractor 524
From the output of the convolutional filter circuit 523, (n−a
i) The output from the function generating circuit 525 is subtracted, and the result is supplied to the divider 526. The divider 526 is adapted to perform inverse convolution processing on the input signal to obtain a masking threshold.

The minimum audible curve generating circuit 532 supplies the data indicating the minimum audible curve to the synthesizing circuit 527. The synthesizing circuit 527 and the data indicating the minimum audible curve output from the minimum audible curve generating circuit 532 and the divider 526.
The masking threshold output from the above is synthesized and output. The subtractor 528 subtracts the output signal from the synthesis circuit 527 and the output from the energy calculation circuit 522 supplied via the delay circuit 529, and outputs the result.

The correction information output circuit 533 outputs data corresponding to a predetermined equal loudness curve. The permissible noise correction circuit 530 corrects the permissible noise level of the output signal from the subtractor 528 based on the output signal from the correction information output circuit 533, and outputs it via the output terminal 531.

In FIG. 6, the input terminal 521 has an MD
The spectrum data in the frequency domain from the CT circuits 402L, 402M, and 402H is supplied.

This frequency domain data is supplied to the energy calculation circuit 522 for each band, and the energy for each critical band (critical band) is, for example, squared for each amplitude value within the band (for example, the peak of the amplitude value). It can be obtained by calculating the sum of the squares of the values. Instead of the energy for each band, a peak value, an average value, etc. of the amplitude value may be used. The spectrum of the total sum value of each band as the output from the energy calculation circuit 522 is generally called a Bark spectrum (SB).
FIG. 7 shows the Bark spectrum for each such critical band. However, in FIG. 7, in order to simplify the illustration, the number of bands of the critical band is represented by 12 bands (bands B1 to B12).

Here, in order to consider the influence of so-called masking of the Bark spectrum, a convolution processing is performed such that the Bark spectrum is multiplied by a predetermined weighting function and added. Therefore, the convolution filter circuit 5 outputs the output of the energy calculation circuit 522 for each band, that is, each value of the Bark spectrum.
23. The convolution filter circuit 523 includes, for example, a plurality of delay elements that sequentially delay input data, and a plurality of multipliers that multiply outputs from these delay elements by a filter coefficient (weighting function) (for example, 25 pieces corresponding to each band). And a sum total adder that sums the outputs of the respective multipliers.

The above-mentioned masking is a phenomenon in which one signal is masked by another signal and becomes inaudible due to human auditory characteristics. The masking effect is due to the time domain audio signal. There are an axial masking effect and a simultaneous time masking effect by a signal in the frequency domain. Due to these masking effects, even if there is noise in the masked portion, this noise will not be heard. For this reason, in an actual audio signal, noise within the masked range is regarded as acceptable noise.

Here, a specific example of the multiplication coefficient (filter coefficient) of each multiplier of the convolution filter circuit 523 will be described. When the coefficient of the multiplier M corresponding to an arbitrary band is 1, the multiplier M-1 gives a coefficient of 0.15, multiplier M-2 gives a coefficient of 0.0019, and multiplier M-3 gives a coefficient of 0.0000.
086, multiplier M + 1 gives a coefficient of 0.4, multiplier M + 2
By multiplying the output of each delay element by a coefficient of 0.06 with a coefficient of 0.007 by a multiplier M + 3, the convolution processing of the Bark spectrum is performed. However, M is 1 to 2
It is an arbitrary integer of 5.

Next, the output of the convolution filter circuit 523 is supplied to the subtractor 524. The subtractor 524 calculates a level α corresponding to an allowable noise level (allowable noise level) described later in the convoluted area. It should be noted that the level α corresponding to this allowable noise level is a level that becomes an allowable noise level for each band of the critical band by performing inverse convolution processing, as described later. here,
The subtractor 524 is supplied with an allowance function (function expressing a masking level) for obtaining the level α. The level α is controlled by increasing or decreasing this allowance function. The allowance function is supplied from the (n-ai) function generating circuit 525 as described below.

That is, the level α corresponding to the allowable noise level can be obtained by the following equation, where i is the number given in order from the low band of the critical band.

Α = S- (n-ai)

In this equation, n and a are constants, and a
> 0 and S are the intensities of the bark spectrum subjected to the convolution processing, and (n-ai) in the formula is the tolerance function. N as an example
= 38, a = -0.5 can be used.

In this way, the level α is obtained, and this data is supplied to the divider 526. The divider 526 is for deconvoluting the level α in the convolved region. Therefore, by performing this inverse convolution processing, the masking threshold can be obtained from the level α.
That is, this masking threshold becomes the allowable noise spectrum. Although the inverse convolution processing requires a complicated calculation, the inverse convolution is performed using the simplified divider 526 in this embodiment.

Next, the masking threshold is supplied to the subtractor 528 via the synthesizing circuit 527. Here, the output from the energy detection circuit 522 for each band, that is, the above-described Bark spectrum (SB) is supplied to the subtractor 528 via the delay circuit 529. Therefore, the subtractor 528 performs a subtraction operation on the masking threshold and the Bark spectrum, so that the Bark spectrum is equal to or lower than the level indicated by the masking threshold (MS) level, as shown in FIG. Will be masked. The delay circuit 529 is provided to delay the Bark spectrum from the energy calculation circuit 522 in consideration of the delay amount in each circuit before the combining circuit 527.

The output from the subtracter 528 is taken out via the allowable noise correction circuit 530 and further via the output terminal 531. For example, RO in which the distribution bit number information is stored in advance.
It is supplied to M and the like (not shown). This ROM or the like outputs the energy obtained from the subtraction circuit 528 through the allowable noise correction circuit 530 (the energy of each band and the synthesis circuit 5).
The distribution bit number information for each band is output according to the level of the difference from the output of 27).

The distribution bit number information thus obtained is supplied to the requantizer 406 shown in FIG. 5, so that the MDCT circuit 404 in the requantizer 406.
Each spectrum data in the frequency domain from L, 404M, and 404H is requantized with the number of bits assigned to each band.

That is, in summary, the requantizer 406
Then, in each band band (critical band) of the critical band or in the high band, depending on the level of the difference between the energy or peak value of the band obtained by further dividing the critical band into a plurality of bands and the output of the synthesizing circuit 527. The spectrum data for each band is quantized with the allocated number of bits.

By the way, at the time of synthesizing by the above-mentioned synthesizing circuit 527, data indicating a so-called minimum audible curve (RC) which is the human auditory characteristic as shown in FIG. 8 supplied from the minimum audible curve generating circuit 532. And the masking threshold (MS) can be combined.
In this minimum audible curve, if the absolute noise level is below the minimum audible curve, no noise will be heard.

Even if the coding (coding method) is the same, the minimum audible curve is different due to the difference in the reproduction volume at the time of reproduction. However, in a realistic digital system, for example, music to a 16-bit dynamic range is generated. Since there is not much difference in how to go, for example 4kH
If the quantization noise in the most audible frequency band around z is not heard, it is considered that the quantization noise below the level of this minimum audible curve is not heard in other frequency bands.

Therefore, assuming that the system is used in such a manner that noise near the dynamic range of the system of 4 kHz cannot be heard, the minimum audible curve (R
If the allowable noise level is obtained by synthesizing C) and the masking threshold (MS) together, the allowable noise level in this case can be up to the shaded portion in FIG. Become. In this embodiment, the level of 4 kHz of the minimum audible curve is set to the minimum level equivalent to 20 bits, for example. Further, FIG. 8 also shows the signal spectrum (SS) at the same time.

In the allowable noise correction circuit 530,
The allowable noise level in the output from the subtractor 528 is corrected based on the information on the equal loudness curve supplied from the correction information output circuit 533, for example. here,
The equal loudness curve is a characteristic curve relating to human auditory characteristics, and is obtained by, for example, obtaining the sound pressure of sound at each frequency heard at the same loudness as a pure tone of 1 kHz and connecting the curves, and the equal sensitivity curve of loudness ( Isoloudness curve) is also called. Further, this equal loudness curve draws a curve substantially the same as the minimum audible curve (RC) shown in FIG.

In this equal loudness curve, for example, in the vicinity of 4 kHz, the sound pressure is 8 to 1 at 1 kHz.
Even if it drops by 0 dB, it sounds the same as 1 kHz. Conversely, the sound pressure around 50 Hz is about 15 dB lower than the sound pressure at 1 kHz.
If it is not high, it will not sound the same size. For this reason, noise that exceeds the level of the minimum audible curve (allowable noise level)
It is understood that it is better to have a frequency characteristic given by a curve corresponding to the equal loudness curve. From this, it can be seen that correcting the allowable noise level in consideration of the equal loudness curve is suitable for human auditory characteristics.

Next, FIG. 9 shows a concrete configuration example of the decoder 133a of FIG. 3 corresponding to the encoder 105a of FIG.
Since the configurations and operations of the decoders 133b to 133e are basically the same as those of the decoder 133a,
Here, the description is omitted.

That is, the decoder shown in FIG. 9 is, for example, for decoding a coded signal for one channel out of each channel read from a motion picture film by a magnetic head or an optical head. .

In FIG. 9, the encoded data from the demultiplexer 132 shown in FIG. 3 is supplied to the input terminal 601, and this is supplied to the deformatter 602. The deformatter 602 performs the reverse process of the process executed by the formatter 407 shown in FIG. 5, and each parameter information of each unit and the requantized spectrum signal (that is, the quantized MDCT).
Coefficient) is obtained.

The quantized MDCT coefficients for each unit from the deformatter 602 are respectively supplied to the low frequency decoding circuit 603L, the middle frequency decoding circuit 603M, and the high frequency decoding circuit 603H. Supplied. Also,
These decoding circuits 603L, 603M, and 603H
Is also given parameter information from the deformatter 602. Each decoding circuit 603L, 603M, and 60
3H uses this parameter information to cancel bit allocation and perform decoding.

The outputs of these decoding circuits 603L to 603H are supplied to the corresponding IMDCT (inverse MDCT) circuits 604L to 604H. Also, each IMD
The parameter information is also supplied to the CT circuits 604L to 604H, where the frequency domain signal is converted into the time domain signal. The time domain signals of these subbands are
The band synthesis circuit 605 decodes into a full band signal,
It is output from the output terminal 606.

The data encoded by the encoding device as shown in FIG. 1 is a DVD (Digital Video Disc), a CD-ROM (Compact Disc: Read Only Memory), or an MD (Mini-disc). It is possible to record on a recording medium such as a disc: MiniDisc. Therefore, it becomes possible to record the data that has been encoded with high efficiency by the encoding device in a recording medium such as a DVD and reproduce it by a decoding device such as a video disc player.

As described above, the compression rate can be increased by using the mixed processed data of one or a plurality of channels, which is obtained by mixing a part or all of the digital signals of a plurality of channels, as the reproduction data. In addition, depending on the characteristics of the audio signal and the playback environment, the channel to which the mixed processing data used as the reproduction data belongs,
By changing the usage rate in units of frames, it is possible to suppress changes in the sound field when the original audio signal is reproduced.

Further, in accordance with the characteristics of the digital signals of the frames both before and after the frame, the channel to which the mixed processing data used as the reproduction data belongs and the ratio of use thereof are changed, or the original digital signal is recorded. By changing the frequency band that is not used, it is possible to avoid an unstable reproduced sound field due to a sudden change in the processing method.

Furthermore, by using a part or all of the mixed processing data of one or a plurality of channels obtained by mixing a part or all of the digital signals of a plurality of channels, in a frame unit of the method of reproducing the original digital signal. By responding to the change of, it is possible to suppress the change of the sound field at the time of reproduction.

The above-described coding method and decoding method of the present invention can be applied not only to the so-called ATRAC method used in the embodiments but also to other coding methods. In particular, the effect is high in a coding method in which frequency information is converted by orthogonal conversion.

Further, although cases have been described with the above embodiments where the audio data of a plurality of channels are encoded or decoded, the invention is not limited to audio data.

In each of the above embodiments, the number of mixed channels is two, but the number is not limited to this and any number of channels can be used. In that case, the configuration of each of the above-described embodiments is changed in accordance with the number of channels subjected to the mixing process.

[0124]

According to the encoding method of the first aspect and the encoding apparatus of the second aspect, one of the digital signals of at least one channel is selected according to the characteristics of the digital signal and the reproduction environment. Partially or completely frequency bands are mixed into at least one mixing channel, and the individually coded individually coded digital signals are extracted from the digital signal, excluding the signal reproduced by the mixed digital signals of the mixing channels, and mixed. Since the digital signal and the individually encoded digital signal are encoded, the original digital signal can be reproduced based on the mixed digital signal, and the compression rate of the digital signal can be increased. Further, by changing the frequency band of the original digital signal restored by the mixed digital signal for each frame, it is possible to suppress the change of the sound field at the time of reproduction.

According to the decoding method of the eleventh aspect and the decoding apparatus of the twelfth aspect, the mixed digital signal is decoded based on the encoded information, and the decoded mixed digital signal The restoration data for restoring the digital signal is created using part or all of the
Individually encoded digital signal and restoration data are combined,
Since the digital signal is restored, the original digital signal can be restored based on the mixed digital signal. Also, by changing the method of restoring the original digital signal using the mixed digital signal for each frame, it is possible to suppress the change in the sound field due to the restoration.

According to the recording medium of the eighteenth aspect, since the mixed digital signal and the individually coded digital signal coded based on the predetermined coded information are recorded, the high efficiency coding is performed. The recorded digital signal can be recorded and reproduced. Further, it becomes possible to provide a more stable reproduced sound field.

[Brief description of the drawings]

FIG. 1 is a block diagram showing a configuration of an embodiment of an encoding device of the present invention.

FIG. 2 is a block diagram showing a configuration example of a processed data extractor of FIG.

FIG. 3 is a block diagram showing the configuration of an embodiment of a decoding device of the present invention.

FIG. 4 is a block diagram showing a configuration example of a combined data generator of FIG.

5 is a block diagram showing a configuration example of the encoder of FIG. 1. FIG.

6 is a block diagram showing a configuration example of a bit allocator that constitutes the encoder of FIG.

FIG. 7 is a diagram for explaining a Bark spectrum (SB) and a masking threshold level (MS).

FIG. 8: Signal level (SS), minimum audible curve (R
It is a figure showing C) and a masking threshold (MS).

9 is a block diagram showing a configuration example of the decoder of FIG.

FIG. 10 is a block diagram showing a configuration example of a multi-channel audio encoding device when the present invention is not used.

FIG. 11 is a block diagram showing a configuration example of a multi-channel audio decoding device when the present invention is not used.

FIG. 12 is a block diagram showing a configuration example of a 2-channel audio decoding device in the case where the present invention is not used.

[Explanation of symbols]

 101 Input Terminal 102 Mixer 103 Processed Data Extractor 105 Encoder 106 Multiplexer 107 Output Terminal 131 Input Terminal 132 Demultiplexer 133 Decoder 134 Synthetic Data Generator 135 Synthesizer 136 Output Terminal 201 Input Terminal 202 Band Division Filter 203 Parameter Recording Memory 204 Processed data analyzer 205 Processed data generator 206 Output terminal 211 Input terminal 212 Band division filter 213 Parameter analyzer 214 Reproduction data generator 215 Reproduction data synthesizer 216 Output terminal 401 Band division filter 402L Low band MDCT circuit 402M Medium band MDCT Circuit 402H High-frequency MDCT circuit 403 Block size evaluator 404L, 404M, 404H Normalization circuit 405 Bit distributor 406 Requantizer 407 Formatter 424 Input terminal 425 Output terminal 521 Input terminal 522 Energy calculation circuit for each band 523 Convolution filter circuit 524 Subtractor 525 (n-ai) function generation circuit 526 Divider 527 Synthesis circuit 528 Subtractor 529 Delay circuit 530 Allowable Noise correction circuit 531 Output terminal 532 Minimum audible curve generation circuit 533 Correction information output circuit 601 Input terminal 602 Deformatter 603L, 603M, 603H Decoding circuit 604L, 604M, 604H IMDCT circuit 605 Band synthesis circuit 606 Output terminal

──────────────────────────────────────────────────続 き Continued on the front page (51) Int.Cl. ⁶ Identification code Agency reference number FI Technical display location H04N 7/24 H04N 7/13 Z

Claims (18)

[Claims] 1. An encoding method for encoding a digital signal of a plurality of frames of a plurality of channels, comprising: corresponding to a characteristic of the digital signal and a reproduction environment,
A frequency band of a part or the whole of the digital signal of at least one of the channels is mixed into at least one mixing channel, and the digital signal except the signal reproduced by the mixing digital signal of the mixing channel is individually coded. An encoding method comprising extracting an individual encoded digital signal to be encoded and encoding the mixed digital signal and the individual encoded digital signal. 2. An encoding device for encoding digital signals of a plurality of frames of a plurality of channels, wherein:
Mixing means for mixing part or all of the frequency band of the digital signal of at least one of the channels into at least one mixing channel; and excluding the signal reproduced from the mixed digital signal of the mixing channel from the digital signal, An encoding device comprising: an extraction unit that extracts an individually encoded digital signal to be individually encoded, and an encoding unit that encodes the mixed digital signal and the individually encoded digital signal. 3. The individually coded digital signal, wherein the extraction means sets the mixing channel used for reproducing the digital signal in the frame unit, and excludes the mixing digital signal of the set mixing channel. The coding device according to claim 2, wherein the signal is extracted. 4. The extraction unit changes the ratio of the mixed channel to which the mixed digital signal used for reproducing the digital signal belongs, in units of the frames. Encoding device. 5. The extracting means changes the frequency band, in which the original digital signal is not encoded, in units of the frame, corresponding to the mixed digital signal used to reproduce the digital signal. The encoding device according to claim 2. 6. The encoding device according to claim 2, wherein the characteristic of the digital signal is a frequency characteristic. 7. The encoding device according to claim 2, wherein the characteristic of the digital signal is an amplitude characteristic. 8. The characteristic of the digital signal includes the characteristic of the digital signal of another frame before or after a predetermined one of the frame to be encoded, or both before and after. Item 2. The encoding device according to item 2. 9. The encoding device according to claim 2, wherein the digital signal is a digital audio signal. 10. The encoding apparatus according to claim 2, wherein the digital signal is a frequency component obtained by converting a time axis into a frequency axis. 11. A part or all of frequency bands of a digital signal of at least one channel of a plurality of channels are mixed in at least one mixing channel, and corresponding to a frequency characteristic of the digital signal and a reproduction environment,
An individual coded digital signal to be individually coded is extracted from the digital signal excluding a signal reproduced by the mixed digital signal of the mixed channel, and the mixed digital signal coded based on predetermined coding information. And a decoding method for decoding the individual coded digital signal, wherein the mixed digital signal is decoded based on the coded information, and a part or all of the decoded mixed digital signal is used. A decoding method comprising: creating restoration data for restoring the digital signal, synthesizing the individually encoded digital signal and the restoration data, and restoring the digital signal. 12. A part or all of frequency bands of digital signals of at least one channel of a plurality of channels are mixed in at least one mixing channel, and corresponding to frequency characteristics and reproduction environment of the digital signals,
An individual coded digital signal to be individually coded is extracted from the digital signal excluding a signal reproduced by the mixed digital signal of the mixed channel, and the mixed digital signal coded based on predetermined coding information. And a decoding device for decoding the individual coded digital signal, wherein the decoding means decodes the mixed digital signal based on the coded information, and the mixing means decoded by the decoding means. Restoration data creating means for creating restoration data for restoring the digital signal using part or all of the digital signal, the restoration data created by the restoration data creating means, and the individual Recovery means for synthesizing the encoded digital signals and restoring the digital signals. Apparatus. 13. The restoration data creating means changes the mixed channel to which the mixed digital signal used when creating the restoration data belongs, on a frame-by-frame basis. Decoding device. 14. The restoration data creating means changes the proportion of the mixed channel to which the mixed digital signal used when creating the restoration data belongs in the frame unit. Decoding device as described. 15. The decoding device according to claim 12, wherein a frequency band of the digital signal restored by the restoration data is changed in the frame unit. 16. The encoding device according to claim 12, wherein the digital signal is a digital audio signal. 17. The encoding apparatus according to claim 12, wherein the digital signal is a frequency component obtained by converting a time axis into a frequency axis. 18. A part or all of the frequency bands of digital signals of at least one channel of a plurality of channels are mixed in at least one mixing channel, and the digital signals are generated according to the frequency characteristics and reproduction environment of the digital signals. An individual coded digital signal to be individually coded is extracted from a signal excluding a signal reproduced by the mixed digital signal of the mixed channel, and the mixed digital signal coded based on predetermined coding information and the A recording medium on which an individually encoded digital signal is recorded.