JP3227942B2 - High efficiency coding device - Google Patents

Description

DETAILED DESCRIPTION OF THE INVENTION

[0001]

BACKGROUND OF THE INVENTION 1. Field of the Invention The present invention relates to a high-efficiency compression / encoding of a multi-channel digital signal used in a stereo such as a movie film projection system, a video tape recorder, a video disk player, etc., or a so-called multi-surround sound system. The present invention relates to an encoding device.

[0002]

2. Description of the Related Art There are various methods and apparatuses for high-efficiency encoding of signals such as audio or voice. For example, a time domain audio signal or the like is divided into blocks for each unit time, and a time axis signal for each block is obtained. Is transformed into a signal on the frequency axis (orthogonal transform), divided into a plurality of frequency bands, and is encoded in each band. Band division coding (sub-band coding: SBC), which is a non-blocking frequency band division method in which an audio signal or the like is divided into a plurality of frequency bands and encoded without being blocked for each unit time, can be used. .
Further, a high-efficiency coding method and apparatus combining the above-described band division coding and transform coding are also considered.In this case, for example, after performing band division by the above band division coding, The signal for each band is orthogonally transformed into a signal in the frequency domain, and the orthogonally transformed band is coded.

[0003] Here, as a filter for band division in the above-mentioned band division coding, for example, there is a filter such as QMF, which is a 1976 RECrochiere Digital coding of sp.
eechin subbands Bell Syst.Tech. J. Vol.55, No.8 1
976 states. Also, ICASSP 83, BOSTON Po
lyphase Quadrature filters-A new subband codingtec
hnique Joseph H. Rothweiler describes an equal bandwidth filter splitting method and apparatus.

As the above-described orthogonal transform, for example, an input audio signal is divided into blocks in a predetermined unit time (frame), and a fast Fourier transform (FF) is performed for each block.
T), cosine transform (DCT), modified DCT
There is an orthogonal transformation that transforms a time axis into a frequency axis by performing transformation (MDCT) or the like. About this MDCT, ICASSP 1987 Subband / Transform Coding Using Fil
ter Bank Designs Basedon Time Domain Aliasing Canc
ellation JPPrincen ABBradley Univ. of Surrey R
oyal Melbourne Inst. of Tech.

Further, as a frequency division width when quantizing each frequency component divided into frequency bands, for example, there is a band division in consideration of human auditory characteristics. That is, an audio signal may be divided into a plurality of bands (for example, 25 bands) with a bandwidth generally called a critical band (critical band) such that the bandwidth becomes wider as the band becomes higher. When encoding data for each band at this time, predetermined bits are allocated to each band, or encoding is performed by adaptive bit allocation to each band. For example, when the coefficient data obtained by the MDCT process is encoded by the bit allocation, the M
Encoding is performed on the MDCT coefficient data for each band obtained by the DCT process with an adaptive number of allocated bits.

The following two methods and devices are known as the above-mentioned bit allocation method and a device therefor. IEEE Tra
nsactions of Accoustics, Speech, and Signal Processi
In ng, vol. ASSP-25, No. 4, August 1977, bits are allocated based on the magnitude of the signal in each band. Also,
ICASSP 1980 The critical band coder--digital encod
ing ofthe perceptual requirements of the auditory
system MA Kransner MIT describes a method and an apparatus for obtaining a required signal-to-noise ratio for each band and performing fixed bit allocation by using auditory masking.

[0007]

By the way, in a high-efficiency compression coding method of an audio signal using, for example, the above-described sub-band coding or the like, the audio data is reduced to about 1 by utilizing the characteristics of human hearing. A method of compression to / 5 has already been put to practical use. As a method of compressing the audio data to about 1/5, for example, a so-called ATRAC (Adaptive TRansform Acousti
c Coding).

In addition to ordinary audio equipment, for example, in a stereo or multi-surround sound system such as a movie film projection system, a high-definition television, a video tape recorder, a video disk player, etc., for example, 4 to 8 channels, etc. Of multi-channel audio or voice signals,
Even in this case, it is desired to perform high-efficiency coding for reducing the bit rate.

In particular, digital audio signals of, for example, eight channels of a left channel, a left center channel, a center channel, a right center channel, a right channel, a surround left channel, a surround right channel, and a subwoofer channel are recorded on the movie film. In such a case, high-efficiency coding for reducing the bit rate is required. That is, for example, an area capable of recording the above eight channels of 16-bit linearly quantized audio data at a sampling frequency of 44.1 kHz as used in a so-called CD (compact disk) is secured on the movie film. Therefore, it is necessary to compress the audio data.

[0010] It should be noted that the 8
Each channel of the channels includes, for example, a left speaker, a left center speaker, a center speaker, a right center speaker, a right speaker, and a surround left disposed on a screen side on which an image reproduced from an image recording area of the movie film is projected by a projector. They correspond to speakers, surround light speakers, and subwoofer speakers, respectively. Here, the center speaker is arranged at the center on the screen side, and outputs a reproduction sound based on the audio data of the center channel, and outputs the most important reproduction sound such as an actor's dialogue. The subwoofer speaker outputs a reproduced sound based on the audio data of the subwoofer channel. For example, the subwoofer speaker effectively outputs a sound that is perceived as vibration rather than a low-frequency sound such as an explosion sound. It is often used effectively for such purposes. The left speaker and the right speaker are arranged on the left and right of the screen, and output a reproduction sound based on left channel audio data and a reproduction sound based on right channel audio data, and exhibit a stereo sound effect.
The left center speaker is arranged between the left speaker and the center speaker, and the right center speaker is arranged between the center speaker and the right speaker. The left center speaker outputs a reproduced sound based on the left center channel audio data, and the right center speaker outputs a reproduced sound based on the right center channel audio data. Fulfill. Particularly in a movie theater with a large screen and a large capacity, the localization of the sound image tends to be unstable depending on the position of the seats. However, by adding the left center speaker and the right center speaker, a more realistic localization of the sound image is created. It is very effective. Further, the surround left speaker and the surround right speaker are arranged so as to surround the audience seat, and output a reproduction sound by the audio data of the surround left channel and a reproduction sound by the audio data of the surround right channel. It has the effect of giving the impression of shouting. Thereby, a more three-dimensional sound image can be created.

Further, since the surface of a medium such as a film is apt to be scratched or the like, if digital data is recorded as it is, the lack of data is so severe that it is not practical. For this reason, the capability of the error correction code becomes very important, and the data compression needs to be performed to such an extent that the data including the correction code can be recorded in the recording area on the film.

For this reason, as a compression method for compressing the digital audio data of each of the above-mentioned eight channels, an optimal bit allocation is performed in consideration of the characteristics of human hearing as described above. The above-described high-efficiency encoding method (which compresses 16-bit digital audio data at a sampling frequency of 44.1 kHz, which is recorded on a so-called CD (compact disk) or the like, to about 1/5 and achieves a sound quality comparable to a CD) ( A so-called ATRAC method) is applied.

However, the ATRAC method is a coding method for a single channel, and when multi-channel audio data is coded by using the ATRAC method, data dependence between channels, data characteristics of each channel,
Effective data encoding processing using elements such as format characteristics cannot be performed.

In view of the above, the present invention provides a high-efficiency code which can realize a higher compression rate in compression coding of a multi-channel signal and which can use an existing decoding device. It is intended to provide a chemical conversion device.

[0015]

SUMMARY OF THE INVENTION A high-efficiency coding apparatus according to the present invention has been proposed to achieve the above-mentioned object, and normalizes and codes a plurality of channels of audio digital signals. And a scale factor calculating means for calculating a scale factor for each unit of a two-dimensional area of time and frequency for each channel in a high-efficiency coding apparatus which outputs parameter information of coding including at least the scale factor of the normalization together with the signal. ,
Common channel selecting means for selecting at least two or more channels for commonizing the scale factor based on the scale factor, and commonizing the scale factor of the selected channel based on the output of the common channel selecting means And a scale factor sharing means for reducing the parameter information to be output, and increasing the compression ratio by reducing the parameter information.

Here, the common channel selection means calculates a predetermined evaluation function relating to the scale factor of the two channels,
A channel to be shared is selected by applying to a combination of all channels including the already selected channel. Further, the high-efficiency encoding device may include a parameter information sharing unit that shares some or all of the parameter information other than the scale factor for the channel selected by the common channel selection unit. Further, the scale factor sharing means shares the parameter information for each band obtained by dividing the audio digital signal of each channel into a plurality of bands, thereby enabling encoding in accordance with the frequency characteristics of the audio digital signal.

[0017]

According to the present invention, when the signals of the channels, which are not completely identical but are similar to each other, are strongly correlated between the digital signals of the respective channels, the signals of the respective channels are encoded by the existing single channel coding method. Parameter information indicating the characteristics of the data in the encoded data tends to match between the channels. Therefore, by sharing the matching parameter information among the channels, redundancy is suppressed and By discriminating channels having similar parameter information, reduction in sound quality due to commonality is suppressed.

[0018]

Embodiments of the present invention will be described below with reference to the drawings. FIG. 1 shows a configuration for performing the so-called ATRAC coding which is applied to the high-efficiency coding apparatus according to the embodiment of the present invention.

In FIG. 1, the sampled and quantized audio data supplied via the input terminal 24 is firstly subjected to 0 to 5.5 by the band division filter 401.
kHz low band, 5.5kHz-11kHz middle band,
It is divided into three frequency bands of 11 kHz or more (11 kHz to 22 kHz).

Of the signals in these three frequency bands, the low-frequency signal from the band division filter 401 is MD
An MDCT circuit 402L that performs a CT (Modified Discrete Cosine Transform) operation, and an MDCT circuit 402 that performs the same MDCT operation on a mid-range signal.
M and the high-frequency signal are sent to the MDCT circuit 402H, where they are decomposed into frequency components by the MDCT circuits 402L to 402H. At this time, the time block length at the time of performing the MDCT is variable for each frequency band. In a portion where the signal changes abruptly, the time block length is shortened to increase the time resolution, and the signal becomes stationary. In the part, the time block length is lengthened to control the effective transmission of the signal component and the quantization noise.

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 sent to the block size estimator 403, and the block size evaluator 403 determines the time block length of the MDCT, and determines the determined time block length. Information indicating the above M
It is sent to the DCT circuits 402L to 402H.

Note that, of the two types of time block lengths in the MDCT, a long time block length is called a long mode, and corresponds to a time of 11.6 ms. Further, a short time block length is called a short mode, and a high frequency (11 kHz) is used.
z or more) up to 1.45 ms, and 2. in the low range (5.5 kHz or less) and the middle range (5.5 kHz to 11 kHz).
The time resolution is increased up to 9 ms.

In this manner, 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 low, middle, and high ranges by normalization circuits 404L to 404H, and standardized for each unit. (Normalized) (a scale factor is determined).

The bit allocation unit 405 analyzes the components of the audio signal using the characteristics of human hearing. The result of this analysis is sent to the requantizer 406 to which signals from each of the normalization circuits 404L to 404H are supplied.

The requantizer 406 determines the degree of precision with which each unit is to be coded based on the analysis result and parameterizes it (determines the word length), and performs requantization. Do.

Finally, the formatter 407 assembles each parameter information of each unit and the requantized spectrum signal into a bit stream according to a predetermined format. The output of the formatter 407 is output from the output terminal 25.

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

FIG. 2 shows how data in the sound frame is recorded. In FIG. 2, one sound frame is composed of 212 bytes, and 44.1 kHz is set here.
At a sampling rate of z, 512 samples and audio reproduction data equivalent to one channel are compression-encoded.

The 212-byte sound frame data includes a block size mode, sub information amount, word length data, scale factor data,
It consists of spectrum data, redundant scale factor byte, redundant word length byte, lower sub information amount, and lower block size mode. Here, in the 212-byte data, a double writing portion for error correction (redundant scale factor byte, redundant word length byte, lower block size mode)
It is included.

The block size mode is data for recording the evaluation result of the block size evaluator 403 in FIG. 1, and its contents are as shown in Table 1.

[0031]

[Table 1]

As can be seen from Table 1, in the long mode, the low frequency and the middle frequency are decomposed into 128 frequency components by the MDCT operation, and the high frequency is decomposed into 256 frequency components. Further, in the short mode, the low band, the middle band, and the high band are each decomposed into 32 frequency components.

As the sub-information amount, three pieces of information, ie, amount 1, amount 2, and amount 3, are recorded. The amount 1 indicates the number of recorded word lengths and scale factors.
Indicates the number of double-written word lengths, and the amount 3 indicates the number of double-written scale factors. The contents are shown in Table 2.

[0034]

[Table 2]

The word length represents the word length of each unit when requantized. The contents are shown in Table 3.

[0036]

[Table 3]

The scale factor represents a normalized value of each unit. The contents are shown in Table 4.

[0038]

[Table 4]

Here, in this embodiment, data according to the ATRAC system of the above eight channels is recorded, for example, as shown in FIG. That is, in order, 20 bytes of ID data,
Each of the data of the channels Ch1 to Ch8 of 212 bytes and the CRC data of the error correction code of 4 bytes are recorded.

When the encoding process in this case is applied to a specific configuration, as shown in FIG. 4, multi-channel audio data (8-channel data) via input terminals 31 to 38 is assigned to each channel. Are respectively encoded by the encoders 41 to 48 provided, and the resultant values are combined by the multiplexer 49 without any change.
This is a process of outputting from the terminal 50 as one data.

When, for example, stereo data has a strong correlation between data of two channels and similar but not identical data is input, the data is encoded by the encoder for each channel. Of the data
The parameter portion representing the characteristics of data tends to match between these channels. That is, it is redundant to record all such matching parameters equally with the parameters of the other channels. Therefore,
For example, it is appropriate to take measures such as representing the matching part as one channel.

In view of these circumstances, the high-efficiency coding apparatus according to the embodiment of the present invention takes the above circumstances into consideration and suppresses redundancy by sharing parameters of a plurality of channels, and discriminates channels having similar parameters. Reduction of sound quality due to commonality is suppressed.

Hereinafter, the common use of the above parameters will be described.

Here, the only requirement between channels that share parameters is that the block size mode of both channels be the same. However, this is not always the case when it is necessary to use a common memory for reasons such as storage capacity. At that time, the time block length gives priority to the short mode, and if any channel is in the short mode, the common block size mode is also set to the short mode.

If only the processing from the determination of the block size mode to the requantization in FIG. 1 is extracted, and the processing is further divided into small pieces and applied to a specific configuration, the processing can be expressed as a configuration as shown in FIG. it can.

That is, in FIG. 5, audio data of each band from the band division filter 401 is supplied to the terminal 9.
0, and this data is sent to the block size mode determination circuit 91, and the block size mode determination circuit 91 determines the block size mode. After that, the scale factor determining circuit 93 normalizes (determines the scale factor) for each of the units. The next sub-information amount determination circuit 95 determines the number of the above-mentioned scale factors and word lengths to be recorded on the recording medium (sub-information amount determination), and determines the word length by bit allocation based on auditory characteristics. The requantizer 98 requantizes the audio data of each band using the determined word length.

Here, when sharing the parameters,
Since it is impossible to share channels having completely different parameters, channels having the most similar parameters will be shared until the limitations such as the data size and recording capacity of all channels can be avoided. Is preferred. The value used for the evaluation that the parameters are similar, that is, the commonization cost can be obtained by Expression (1). Equation (1) is an equation in the case of using a common cost between two channels. Also, this equation (1)
In the formulas, SF1 (i) and SF2 (i) are two-channel units i (UNIs) for calculating the common cost.
The scale factor value of Ti) is shown.

[0048]

(Equation 1)

For the combination of all channels including the channels whose parameters have already been shared, the sharing cost is calculated using the above equation (1), and the smallest channel is shared. The operation up to this point is continued until it becomes unnecessary to share channels.

When this common cost calculation is applied to a specific configuration, it becomes as shown in FIG. That is, in FIG. 6, data of two channels is supplied to the terminals 100a and 100b, and the block size mode is determined by the block size mode determination circuits 101a and 101b for each channel. The block size mode determination circuits 101a and 101b output block size mode information together with audio data for each block, and the block size mode information is sent to the same check circuit 102.

The same check circuit 102 checks whether the block size mode is the same, and compares the result with the scale factor determining circuit to which the audio data for each block of the corresponding channel is supplied from each of the block size mode determining circuits 101a and 101b. 103a, 10
3b. Each scale factor determination circuit 103
The values of the scale factors from a and 103b are sent to the common cost calculation circuit 104, where the common cost of the above equation (1) is calculated.

The calculation result of each channel from the common cost calculation circuit 104 is output via terminals 111a and 111b, and the audio data of each channel from each scale factor determination circuit 103 is output from terminals 110a and 11b.
0b.

After the channel for sharing the parameter is determined as described above, the processing up to the requantization is performed thereafter. Note that the processing in the case of a channel that does not share parameters remains the processing of FIG. On the other hand, in the case of channels that share parameters,
As shown in the specific configuration of FIG. 7, requantization processing is performed using the common parameters.

That is, in FIG.
a, 113b correspond to the corresponding terminals 111a, 1 in FIG.
11b are supplied, and the terminals 112a, 112b
Are supplied with the outputs of the corresponding terminals 110a and 110b in FIG. A signal passed through the terminals 110a and 110b is supplied to a requantizer 108 provided for each channel.
a, 108b.

On the other hand, the signal via the terminals 113a and 113b is sent to the scale factor sharing circuit 105. The scale factor sharing circuit 105 shares the scale factor. For this commonization, max (SF1, SF2) is obtained for each unit, and this is used as the scale factor of each unit. The above-mentioned max means the maximum value of both scale factors. By choosing the maximum value,
Avoid occurrence of data that cannot be requantized.

The next common sub-information amount determining circuit 106 determines the common sub-information amount, and based on this, the next word length determining circuit 107 performs bit allocation processing to determine the word length. Since this process is performed using the block size mode, the amount of sub information, and the scale factor, the process can be performed in common between channels using only the common data. The processing content is the same as the processing shown in FIG.

Finally, the requantizers 108a and 108b perform requantization for each channel using the obtained common parameters. As a result, encoded data of each channel with common parameters is created and output from terminals 114a and 114b corresponding to each channel.

When the above-described processing is applied to the configuration for encoding the eight-channel audio data, the overall encoding processing is as shown in FIG. 8 as compared with FIG.

That is, in FIG. 8, audio data from the input terminals 31 to 38 of each channel is sent to a common channel encoder 51, and the common channel encoder 51 performs the above-described common processing and encoding. Done. The encoded data of each channel from the common channel encoder 51 is output from the terminal 53 via the multiplexer 52.

By the way, in the decoding corresponding to the above-mentioned encoding, it is necessary to allocate the above-mentioned common data to each channel before the decoder which decodes each channel. That is, as shown in FIG. 9, for example, a case where the parameters of the channel Ch3 and the channel Ch4 among the eight channels are shared is described as an example.
In the data recording condition (131) on the recording medium as shown in FIG. 9A, common parts are distributed to each as shown in FIG. (A decoder for channel Ch3 and a decoder for channel Ch4).

Here, multi-channel data is generally 1
The data is recorded in a single state, and the decoding device needs to perform processing of allocating the data to the data of each channel. In the case of using an encoding method for performing common processing as in the present embodiment, the decoding device temporarily stores, for example, all channels in an internal memory for one sound frame,
Next, the distributor may be designed so that the corresponding parameter can be distributed to each channel. Further, according to the method of the present embodiment, since the range of commonality is completed within one sound frame, it is not necessary to handle a plurality of sound frames on the time axis of the same channel, and data can be easily selected. is there.

ATRAC data in which parameters are shared by the above processing has a smaller data size per channel, as shown in equation (2). SIA * (4 + 6) +16 [bits] (2)

Here, SIA in the equation (2) indicates the amount of sub information. For example, the maximum value of the SIA in the case of the ATRAC system used in this embodiment is 5
Since it is 2, the data size is reduced by a maximum of 536 bits, that is, 67 bytes. Also, A
In the data in the TRAC method, the parameters are double-written, which can be also used in common. For example, if the size of the parameter double writing area is 26 bytes, the size of one sound frame can be shared by 93 bytes out of 212 bytes, so that it can be compressed to a maximum size of about 56%.

Note that, as described above, when the parameters are shared, a quantization error may occur as compared with the case where the parameters are not shared. However, in general, when the data of the ATRAC method is used as a variable length and the size of the coded data of the present embodiment is reduced to the same size as that of the coding system of the present embodiment, Has a smaller quantization error, and thus shows the effectiveness of the encoding method of this embodiment.

Next, another embodiment of the present invention will be described. For example, if there is room in the recording area of the recording medium, or if it is not allowed to generate quantization noise due to a decrease in accuracy due to the commonization, instead of sharing all parameters, some parameters may be used. It is possible to use a method that makes only one common. In other words, this is a method in which only the three elements of the block size mode, the amount of sub information, and the scale factor are shared. Thereby, although the required recording area on the recording medium is increased as compared with the case of the above-described embodiment,
Since the required recording area is smaller than in the case of the ATRAC method and the generation of quantization noise is suppressed, it is effective for improving sound quality. It should be noted that the method of sharing the three of the block size mode, the amount of sub information, and the word length may be somewhat wasteful due to the characteristics of the ATRAC format.

In addition, a method of sharing two elements, that is, a block size mode and a sub information amount, is also conceivable. It should be noted that the method of sharing these two elements has a size of 2 bytes, so that the effect of improving the compression ratio is not as good as in the above embodiments, but it goes without saying that it is better than the ATRAC method.

As still another method, there is a method of making use of the characteristics of the band division of coding to make the band common among only the low band, the middle band, the high band, or a band obtained by combining them. . In this case, it is necessary to partially share the block size mode and match the amount of sub-information, but this makes it possible to perform encoding using data characteristics and channel characteristics such as sharing a band that is not frequently used,
Effective coding becomes possible.

As described above, according to the high-efficiency coding apparatus of the embodiment of the present invention, it is possible to realize a higher compression rate than the ATRAC method in coding multi-channel audio data. Further, as the number of channels increases, the effect increases. In addition, the encoded data in the apparatus of the present embodiment is ATRAC data in which only the parameters are shared, and the common processing is completed within one sound frame. When the divided data is divided for each channel, decoding can be performed by an existing decoding device only by allocating one common parameter to the corresponding channel.

[0069]

As described above, according to the present invention, in the encoding of multi-channel audio data, a scale factor is determined for each unit of a two-dimensional region of time and frequency for each channel, and the scale factor is determined based on the scale factor. By selecting at least two or more channels having the same factor and sharing the scale factor of the selected channel, it is possible to achieve a higher compression ratio than existing compression coding. Further, as the number of channels increases, the effect increases.

The data encoded by the apparatus of the present invention is the data of the existing compression encoding system in which only the parameters are shared, and the common processing is completed within one frame of the existing compression encoding system. Therefore, when the data collected at the time of recording is divided for each channel, only one common parameter is allocated to the corresponding channel.
Decoding with an existing decoding device becomes possible.

[Brief description of the drawings]

FIG. 1 is a block circuit diagram showing a configuration for performing ATRAC coding applied to a high-efficiency coding apparatus according to an embodiment of the present invention.

FIG. 2 is a diagram for explaining how a disc is recorded in a sound frame.

FIG. 3 is a diagram illustrating a state of recording on a recording medium of data obtained by encoding 8-channel audio data by the ATRAC method;

FIG. 4 is a block circuit diagram showing a configuration for performing a process of collecting 8-channel encoded audio data into one data without changing it.

5 is a block circuit diagram showing, as a more detailed configuration, processing from determination of a block size mode to requantization in FIG.

FIG. 6 is a block circuit diagram showing a specific configuration for obtaining a common cost from audio data of two channels.

FIG. 7 is a block circuit diagram showing a specific configuration for commonizing parameters of audio data of two channels and requantizing them.

FIG. 8 is a block circuit diagram showing an overall configuration of an encoding process for requantizing each channel using a common parameter.

FIG. 9 is a diagram for explaining the operation of the decoding device when decoding data encoded by performing parameter sharing.

[Explanation of symbols]

41 to 48 encoders 49, 52 multiplexer 51 common channel encoders 91, 101a, 101b block size mode determination circuit 93, 103a, 103b scale factor determination circuit 95 ..Sub-information amount determination circuits 97, 107 ... word length determination circuits 98, 108a, 108b, 406 ... requantizers 102 ... identical check circuits 104 ... common cost calculation circuits 105 ... · Scale factor common circuit 106 ··· Common sub-information amount determination circuit 401 ··· Band dividing filter 402L, 402M, 402H ··· MDCT circuit 403 ··· Block size evaluator 404L, 404M and 404H ··· Regular Circuit 405 ··· bit allocation device 407 ··· Omatta

──────────────────────────────────────────────────続 き Continued on the front page (58) Field surveyed (Int.Cl. ⁷ , DB name) H03M 7/30 G11B 20/10

Claims (4)

(57) [Claims] 1. A digital audio signal of a plurality of channels.
The signal is normalized and encoded, and a small
At least the encoding including the normalization scale factor
In a high-efficiency encoding device that outputs parameter information, a unit of a two-dimensional region of time and frequency for each channel is used.
Scale factor meter that calculates the scale factor for each unit
Calculation means and the scale factor based on the scale factor.
Select at least two or more channels that share data
The selected channel is selected based on the output of the common channel selecting means and the output of the common channel selecting means.
Scale to make the above scale factors common for
High, characterized in that it comprises a Rufakuta sharing means
Efficiency coding device. 2. The method according to claim 1, wherein the common channel selecting means comprises two channels.
The predetermined evaluation function for the scale factor of the
Combination of all channels including the channel selected in
Select channels to be shared by applying
2. The high-efficiency encoding apparatus according to claim 1, wherein: 3. The method according to claim 1, wherein said selection is made by said common channel selection means.
Parameters other than the scale factor
Parameter information that shares part or all of the meter information
2. The method according to claim 1, further comprising:
High efficiency coding device. Wherein said scale factor sharing means for high efficiency coding according to claim 1, characterized in that the sharing of the parameter information for each band obtained by dividing the audio digital signals of the channels in a multiband apparatus.