JPH07161140A - Apparatuses and methods for transmission and receiving of digital audio signal - Google Patents

Abstract

PURPOSE:To facilitate compression coding of high tone quality and, further, facilitate compression coding of a high compression factor for an important sound by a method wherein the coding corresponding to a channel is performed with the predetermined compression factor for a signal having a large influence upon the sensation of hearing corresponding to a channel. CONSTITUTION:High sensation compression coding circuits 217-222 which apply compression coding with a first coding system using a first compression factor to the digital audio signals of (m) channels (n>m) which have a larger influence upon the sensation of hearing than the digital audio signals of the other (n-m) channels among the digital audio signals of (n) channels are provided. Further, high compression factor coding circuits 223 and 224 which apply compression coding with a second coding system using a second compression factor which is higher than the first compression factor to the digital audio signals of the (n-m) channels are provided. Then the respective compression coding outputs of both the circuits 217-222 and the circuits 223 and 224 are transmitted.

Description

Detailed Description of the Invention

[0001]

BACKGROUND OF THE INVENTION 1. Field of the Invention The present invention, for example, compresses and transmits a multi-channel digital audio signal used in a stereo such as a movie film projection system, a video tape recorder, a video disc player, or a so-called multi surround sound system. The present invention relates to a digital audio signal transmitting apparatus and a corresponding receiving apparatus.

[0002]

2. Description of the Related Art Not only in the case of ordinary audio equipment, but also in stereo or multi-surround 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. It is becoming increasingly common to handle multiple channels of audio or voice signals.

In particular, for business use, digital audio has become multi-channel, and, for example, equipment that handles 8-channel digital audio signals has become widespread. An example of a device that handles the 8-channel digital audio signal is a movie film projection system. Also, in stereo or multi-surround sound systems such as high-definition televisions, video tape recorders, video disc players, etc.
A plurality of channels of audio or voice signals such as 8 channels are being handled.

In the motion picture film projection system that handles the 8-channel digital audio signals, for the motion picture film, for example, a left channel, a left center channel, a center channel, a right center channel, a right channel, a surround left channel, a surround right. Recording of 8-channel digital audio signals of channels and subwoofer channels is being performed. It should be noted that the above 8 recorded on the movie film
Each channel is, for example, a left speaker, a left center speaker, a center speaker, a right center speaker, a right speaker, a subwoofer speaker, which is arranged on the screen side on which an image reproduced from the image recording area of the movie film is projected by the projector. , A surround left speaker arranged on the left side and a surround right speaker arranged on the right side so as to surround the auditorium.

[0005]

However, in the case of recording the above 8-channel digital audio signals on a motion picture film, the motion picture film has a sampling frequency of 44. Since it is difficult to secure an area in which 16-bit linearly quantized audio data of 1 kHz can be recorded for 8 channels, it is necessary to compress and record the 8 channels of audio data.

Further, since a medium called a film is apt to have scratches on its surface, if digital data is recorded as it is, the data will be seriously lost and it will not be put to practical use. For this reason, the capability of the error correction code becomes very important, and the data compression must be performed to the extent that it can be recorded in the recording area on the film including the correction code.

However, when compression coding is performed, the musical instrument, human voice, or the like changes from the original sound. Therefore, when it is adopted as a recording format of a medium in which faithful reproduction of the original sound is required as in the above-mentioned movie film. For important sounds such as human voice, some means for improving the sound quality is required.

In view of the above, the present invention enables high-quality compression coding for particularly important sounds and also enables high-compression compression coding. It is an object of the present invention to provide a digital audio signal transmission apparatus using high-efficiency coding and a receiving apparatus corresponding to the transmission apparatus.

[0009]

The present invention has been proposed in order to achieve the above object, and a digital audio signal transmitting apparatus of the present invention is provided with a plurality of channels (n channels, n is greater than 3). Is a transmission device that compresses and encodes a digital audio signal of
Among the digital audio signals of the above n channels, for the digital audio signals of m (n> m) channels and having a higher audible influence than the digital audio signals of the other nm channels. The first
Second compression coding means for compressing and coding by the first coding method having a compression rate of 1) and a second compression rate higher than the first compression rate for the digital audio signals of the n-m channels. Second compression encoding means for performing compression encoding with a second encoding method having a compression rate of, and a compression encoding output from the first compression encoding means and the second compression encoding. It is characterized in that the compression encoded output from the means is transmitted together.

In the transmission device of the present invention, the n channels are a center channel, a left channel, a right channel, a surround left channel and a surround right channel, and the n-m channels are surround left channels. Surround light channel. Alternatively, the n channels are a center channel, a left channel, a right channel, a left center channel, a right center channel, a surround left channel and a surround right channel, and the nm channels are a surround left channel and a surround right channel. Is.

Further, in the transmission apparatus of the present invention, the first encoding method divides the input digital audio signal into a plurality of bands, divides the digital audio signal in each band into blocks for a plurality of samples, and sets each block unit. Is a system for adaptively compressing and coding the spectrum component orthogonally transformed in accordance with the auditory characteristic, and the second coding system is related to coefficient information obtained by orthogonally transforming an input digital audio signal for every plural samples and its relation. Get sub information to
This is a method in which the bit allocation amount of each channel is determined according to the energy of each channel, and compression coding is adaptively performed with this bit allocation amount. The above transmission includes recording on a recording medium.

Further, the digital audio signal receiving apparatus of the present invention is a digital audio signal of a plurality of channels (n channels, n is a positive integer greater than 3),
The first code having the first compression ratio for a digital audio signal of m (n> m) channels, which has a higher auditory influence than other digital audio signals of nm channels. The second encoding method has a second compression rate higher than the first compression rate with respect to the digital audio signals of the n−m channels while being compression-encoded by the encoding method. A receiver for a digital audio signal for receiving a compression-encoded signal, wherein the first encoding method is applied to the digital audio signals of the m channels among the digital audio signals of the n channels. Corresponding to the second encoding method for the digital audio signals of the n−m channels and the first decoding means for performing the first decompression / decoding. It is characterized in that a second decoding means for performing a second decompression and decoding.

[0013]

According to the digital audio signal transmission apparatus of the present invention, the first compression rate is less likely to be deteriorated in hearing for a signal of a channel consisting of a signal having a high hearing influence among a plurality of channels. Compression encoding is performed by the first encoding method of No. 2, and a second encoding method of a second compression rate higher than the first compression rate for a signal of a channel composed of a signal having a low auditory influence. The compression coding is performed with.

Further, according to the digital audio signal receiving apparatus of the present invention, the first compression which causes less deterioration in the auditory sense than the signal of the channel consisting of the signals having a high auditory influence among the signals of the plurality of channels. The second compression rate of the second compression rate higher than the first compression rate with respect to the signal of the channel composed of the signal compression-encoded by the first rate encoding method and the signal having a low auditory influence. The signals compressed and encoded by the encoding method are each decoded.

[0015]

Embodiments of the present invention will be described below with reference to the drawings.

FIG. 1 shows the configuration of a digital audio signal transmission apparatus according to the present invention. A digital audio signal transmission apparatus according to an embodiment of the present invention is a transmission apparatus that compresses and encodes digital audio signals of n (n is a positive integer greater than 3; for example, 8 in this embodiment) channels. Is.

Here, in this embodiment, the above n pieces (8 pieces) are used.
The left channel (L
ch), left center channel (LCch), center channel (Cch), subwoofer channel (SWc)
h), right center channel (RCch), right channel (Rch), surround left channel (LBc)
h), a surround right channel (RBch) is used. In the present invention, the n channels are not limited to the above example, but may be, for example, 5 channels of a center channel, a left channel, a right channel, a surround left channel and a surround right channel, or a center channel, a left channel and a right channel. It is also possible to use seven channels including a channel, a left center channel, a right center channel, a surround left channel and a surround right channel.

The digital audio signal transmitting apparatus according to the present embodiment is a digital audio signal of m (n> m, m = 6 in the example of FIG. 1) channels of the above eight channels of digital audio signals. Other nm
(In the example of FIG. 1, mn = 8−6 = 2) For a signal that has a higher auditory influence than a digital audio signal of the number of channels, a compression code is generated by the first encoding method with the first compression ratio. The high-audibility compression coding circuits 217 to 222 as the first compression coding means for converting the digital audio signals into the n-m channels and the second compression higher than the first compression ratio. A high-rate compression coding circuit 223, 224 which is a second compression coding means for performing compression coding by a second coding method having a rate.
It is characterized in that the compression coded outputs from 17 to 222 and the compression coded outputs from the high rate compression coding circuits 223 and 224 are both transmitted.

In the present embodiment, among the eight channels, the channels (six channels) made up of the signals having a high perceptual impact that are compressed by the first coding method are used as the left channel and the left center. A channel, a center channel, a subwoofer channel, a right center channel and a right channel, and channels (nm channels) compressed by the second encoding method are a surround left channel and a surround right channel. Of course, the present invention is not limited to these, and for example, when the above n channels are, for example, 5 channels of a center channel, a left channel, a right channel, a surround left channel and a surround right channel, the m channels are replaced by the center channel. The left channel and the right channel may be three channels, and the n−m channels may be the surround left channel and the surround right channel. Similarly, when the n channels are, for example, 7 channels of a center channel, a left channel, a right channel, a left center channel, a right center channel, a surround left channel and a surround right channel, the m channels are referred to as center channels. A left channel and a right channel may be used, and the above n−m channels may be a surround left channel and a surround right channel.

Further, in the transmission apparatus of the embodiment of the present invention, the above-described sub-band coding or the like is used for the high-audibility compression coding circuits 217 to 222 which perform compression coding by the first coding method. It is a high-efficiency compression encoding method for audio signals, which makes use of human auditory characteristics,
A method of compressing audio data to about ⅕, that is, a spectrum in which an input digital audio signal is divided into a plurality of bands, the digital audio signal of each band is divided into a plurality of samples, and orthogonally transformed in each block. A method of adaptively compression-coding data according to human auditory characteristics (for example, so-called ATRAC: Adap
tive TRansform Acoustic Coding method) is used.

Further, in the high-rate compression coding circuits 223 and 224 which carry out compression coding by the second coding method, coefficient data obtained by orthogonally transforming an input digital audio signal for every plural samples and sub-information related thereto ( The word length information and the scale factor information) are obtained, the bit allocation of each channel is determined according to the energy of each channel, and the compression encoding is adaptively performed by the bit allocation of each channel. Needless to say, the present invention is not limited to each of these encoding methods.

Further, in this embodiment, the above transmission is as follows.
For example, recording on motion picture film, recording on disk-shaped recording media such as optical disks, magneto-optical disks, phase-change optical disks, magnetic disks, tape-shaped recording media such as magnetic tape, recording on semiconductor memory, IC cards, etc. Can be mentioned.

When the transmission is recorded on a motion picture film, it corresponds to a digital surround system in which speakers are arranged as shown in FIG. 2, for example. Each channel is a center (C) channel, a subwoofer (SW) channel, a left (L) channel, a left center (CL) channel, a right (R) channel, a right center (CR) channel, a left surround (LB) channel.
There are eight channels, a light surround (RB) channel.

In FIG. 2, each of the eight channels recorded on the motion picture film has a screen 101 on which an image reproduced from an image recording area of the motion picture film is projected by a projector (projector 100). Left speaker 106, left center speaker 104, center speaker 10 arranged on the side
2, light center speaker 105, light speaker 1
07, surround left speakers 108 and 200, surround right speakers 109 and 201, and subwoofer speaker 103.

The center speaker 102 is arranged in the center of the screen 101 side and outputs a reproduced sound based on the audio data of the center channel, and outputs the most important reproduced sound such as an actor's dialogue. The subwoofer speaker 103 outputs a reproduced sound based on audio data of the subwoofer channel, and effectively outputs a sound that is felt as vibration rather than a low frequency sound such as an explosion sound, and an explosion scene. Is often used effectively. The left speaker 106 and the right speaker 107 are arranged on the left and right sides of the screen 101, and output a reproduced sound by the left channel audio data and a reproduced sound by the right channel audio data, and exhibit a stereo sound effect. The left center speaker 104 and the right center speaker 105 are arranged between the center speaker 102 and the left speaker 106 and the right speaker 107, and are reproduced by the left center channel audio data and reproduced by the right center channel audio data. It outputs sound and plays an auxiliary role for the left speaker 106 and the right speaker 107, respectively. Particularly in a movie theater or the like where the screen 101 is large and the number of persons accommodated is large, the localization of the sound image tends to become unstable depending on the position of the seat.
4 and the light center speaker 107 are added, it is effective in creating a more realistic localization of the sound image. Further, the surround left speaker 108 and the surround right speaker 109 are arranged so as to surround the spectators' seats, and output a reproduced sound by the audio data of the surround left channel and a reproduced sound by the audio data of the surround right channel. It has the effect of giving the impression of being wrapped in clapping and cheering. As a result, a more stereoscopic sound image can be created.

Returning to FIG. 1, the transmission apparatus of FIG. 1 for handling the above-described 8-channel digital audio signal is
In order to obtain the above 8-channel digital audio signals, the microphones 201 to 208 corresponding to the respective channels are obtained.
Corresponding analog audio signals from A /
The D converters 209 to 216 convert the digital audio signals.

The digital audio signals of the respective channels from the respective A / D converters 209 to 216 are sent to the corresponding compression encoding circuits 217 to 224 and compression encoded respectively. Note that each of these compression encoding circuits 217-2
The specific configuration of 24 will be described later.

The above-mentioned high-audibility compression encoding circuits 217 to 222.
The compressed and encoded audio data from are sent to the subsequent stage from the corresponding output terminals 233 to 238.

On the other hand, the high-rate compression coding circuits 223, 2
The compression-encoded audio data from 24 is added by the adder 227 after only the high frequency components are extracted by the high pass filters 225 and 226, respectively. The compression-encoded audio data from the high-rate compression encoding circuits 223 and 224 are also sent to low-pass filters 228 and 229, respectively. The data from the adder 227 is supplied to the low pass filter 2 corresponding to the surround left channel side by the adder 230.
28 output is added and output from the output terminal 231.
The output of the low pass filter 229 corresponding to the surround right channel is output from the output terminal 232.

Here, in the surround left channel and the surround right channel, the outputs from the high pass filters 225 and 226 are added by the adder 227, and the addition output of the adder 227 is further added.
The reason for adding to the output of the low-pass filter 228 of the surround left channel by 8 is as follows.

That is, the human ear has less sense of localization for high-frequency components, and therefore, for a high-frequency component, no matter which speaker from a plurality of speakers, for example, which human speaker It is difficult to hear if it is coming out. Therefore, each high frequency component of the audio signals of a plurality of channels is
Even if you send only to the speaker corresponding to the channel,
It is not perceived by humans that the high frequency component of the audio signal of each channel is output from only the one speaker. Therefore, in this embodiment, as described above,
Of the channels, for example, add the high frequency components of the surround left channel and surround right channel audio,
For example, it is added to the sound of the low frequency component of the surround left channel. As a result, the high frequency components of the two channels, the surround left channel and the surround right channel, can be compressed into one channel.
The crosstalk processing between channels in such a high frequency component is performed not only between the surround left channel and the surround right channel as in the example of FIG. 1 but also between other channels (not shown) (for example, The same can be done for all channels). In this way, it is possible to further increase the compression rate by performing channel crosstalk processing of high frequency components for all channels.

The output terminals 233 to 238 shown in FIG.
, 231 and 232 of each channel are sent to the multiplexer 241 via the corresponding terminals 233 to 238 and 231, 232 of FIG. 3, respectively, and are multiplexed. The output of the multiplexer 241 is transmitted to the transmission line via the output terminal 242 or recorded on the above-mentioned movie film of the present invention, the disc-shaped recording medium, the tape-shaped recording medium, or the like.

Next, the high-hearing-compression encoding circuits 217 to 222 shown in FIG. 1 will be described in detail.

First, FIG. 4 shows the configuration of a specific example of each of the above-described high-audibility compression encoding circuits 217 to 222.

In the compression encoding circuit shown in FIG. 4, the input digital signal is divided into a plurality of frequency bands by a filter or the like, and orthogonal transformation is performed for each frequency band to obtain the spectrum data on the frequency axis. The bits are adaptively allocated and encoded for each so-called critical band (critical band) in consideration of human auditory characteristics described later. At this time, a band obtained by further dividing the critical bandwidth is used in the high band. Of course, the non-blocking frequency division width by a filter or the like may be an equal division width. Further, in this embodiment, the block size (block length) is adaptively changed according to the input signal before the orthogonal transformation, and the critical bandwidth (critical band) is further subdivided in the critical band unit or in the high range. Floating process is performed in the block.

As the band division filter,
For example, there is a filter such as QMF, which is 1976 RECro
chiere Digital coding of speech in subbands Bell
Syst.Tech. J. Vol.55, No.8 1976. Also ICASSP 83, BOSTONPolyphase Quadrature f
ilters-A new subband coding technique Joseph H.Ro
Thweiler describes an equal bandwidth filter partitioning method and apparatus.

Further, the critical band is a frequency band divided in consideration of human auditory characteristics,
It is a band of a certain pure tone when the pure tone is masked by a narrow band noise having the same strength near the frequency of the pure tone. This critical band has a wider bandwidth as it goes higher, and the entire frequency band of 0 to 22 kHz is divided into, for example, 25 critical bands.

That is, in FIG. 4, an audio PCM signal of 0 to 22 kHz, for example, is supplied to the input terminal 10. This input signal is divided into 0 to 11 kHz band and 1 by a band division filter 11 such as so-called QMF.
The signal of 0 to 11 kHz band is divided into the 1 to 22 kHz band and the 0 to 5.5 kHz band and 5.5 to 11 kHz by the band dividing filter 12 such as so-called QMF.
and z-band. 1 from the band division filter 11
A signal in the 1 kHz to 22 kHz band is an MDCT (Modified Discrete Cosine Transform) that is an example of an orthogonal transformation circuit.
5.5 sent from the band division filter 12 to the circuit 13.
A signal in the k to 11 kHz band is sent to the MDCT circuit 14, and a signal in the 0 to 5.5 kHz band from the band division filter 12 is sent to the MDCT circuit 15 to be MDCT processed.

In each MDCT circuit 13, 14, 15 MDCT processing is performed based on the block size determined as described later by the block determination circuits 19, 20, 21 provided for each band.

The MDCT processing in each MDCT circuit 13, 14, 15 is described in ICASSP 1987 Sub.
band / Transform Coding Using Filter Bank Designs B
asedon Time Domain Aliasing Cancellation JPPrinc
en ABBradley Univ. of Surrey Royal Melbourne Ins
t.of Tech.

The information indicating the block size decided by the block decision circuits 19, 20, 21 is sent to the adaptive bit allocation coding circuits 16, 17, 18 described later and output from the output terminals 23, 25, 27. To be done.

Here, the adaptive bit allocation encoding circuit 1
6, 17 and 18, the MDCT circuits 13 and 1
From the outputs of 4 and 15, the energy of each critical band (critical band) or a band obtained by further dividing the critical band in the high band is calculated, for example, the root mean square of each amplitude value in the band is calculated. Sought by
Bit allocation is performed based on the calculation result. Of course, the scale factor itself may be used for subsequent bit allocation. In this case, the calculation of new energy is not required, and the hardware scale is saved. Further, instead of the energy for each band, it is also possible to use the peak value, the average value, etc. of the amplitude values. The spectrum data in the frequency domain or the MDCT coefficient data obtained by the MDCT processing in each MDCT circuit 13, 14, 15 is a so-called critical band (critical band) or a band in which a critical band is further divided in a high band. The data are collected and sent to the adaptive bit allocation coding circuits 16, 17, and 18.

Further, the adaptive bit allocation encoding circuit 1
6, 17, and 18 each spectrum data (or MDCT) according to the block size information and the number of bits assigned to each band obtained by further dividing the critical band (critical band) or the critical band in the high band.
The coefficient data) is requantized (normalized and quantized). Adaptive bit allocation coding circuits 16, 17, 1
The data encoded in 8 are output terminals 22, 24, 26
Taken out through. Further, in the adaptive bit allocation encoding circuits 16, 17, and 18, a scale factor indicating what kind of signal magnitude is normalized and a bit length indicating what kind of bit length is quantized. Information is also sought, and these are also output terminals 22,
It is output from 24 and 26.

The data from the output terminals 22 to 27 are collected and combined into the above-described high-audibility compression encoding circuits 217 to 2 respectively.
22 output.

In the examples of FIGS. 1 and 4, the high-audibility compression encoding circuits 217 to 222 perform bit-wise bit allocation for each channel for compression encoding. It is also possible to perform bit allocation among the encoding circuits 217 to 222 (that is, perform bit allocation among channels corresponding to the circuits 217 to 222).

The high-audibility compression encoding circuits 217 to 222
Each circuit configuration in the case of performing bit allocation between channels will be described below. FIG. 5 shows the configuration of a high-audibility compression encoding circuit that allocates bits among channels. In the configuration of FIG. 5, adaptive bit allocation coding circuits 16, 17, 1
Other components except 8 are basically the same as the corresponding components in FIG.

In the compression encoding circuit shown in FIG. 5, a concrete example of the block size in each MDCT circuit 13, 14, 15 determined by the block determination circuits 19, 20, 21 similar to that in FIG. 4 is shown in FIG. Shown in A and B. 6A shows a case where the orthogonal transform block size is long (orthogonal transform block size in the long mode), FIG.
Shows the case where the orthogonal transform block size is short (orthogonal transform block size in the short mode).
In the specific example of FIG. 6, the three filter outputs are
Each has two orthogonal transform block sizes. That is, for a signal in the low band of 0 to 5.5 kHz and a signal in the mid band of 5.5 to 11 kHz, the number of samples in one block is set to a long block length (A in FIG. 6). When 128 blocks are selected and a short block is selected (B in FIG. 6), the number of samples in one block is a block for every 32 samples. On the other hand, the high side 1
For a signal in the 1 kHz to 22 kHz band, if the block length is long (A in FIG. 6), the number of samples in one block is 2
56 samples and a short block is selected (Fig. 6
In B), the number of samples in one block is a block for every 32 samples. When a short block is selected in this way, the number of samples of orthogonal transform blocks in each band is set to be the same, the time resolution is increased in the higher frequency range, and the number of windows used for blocking is reduced. In addition, the block determination circuits 19, 20, and 21 of the specific example of FIG.
The information indicating the block size determined in step 1 is sent to the adaptive bit allocation coding circuits 16, 17 and 18, which will be described later, and is also output from the output terminals 23, 25 and 27.

In the adaptive bit allocation coding circuits 16, 17 and 18 of the specific example of FIG. 5, the block size information and the critical band (critical band) or the critical band in the high frequency band is further allocated. Re-quantization (normalized and quantized) of each spectrum data (or MDCT coefficient data) according to the number of bits
I am trying to do it. At this time, the adaptive bit allocation coding circuits 16, 17, and 18 adaptively and optimally allocate the used bit amount for each channel by observing the channel bit allocation among the channels, that is, the entire signal of each channel. Allocate simultaneously. The channel bit allocation in this case is performed based on the channel bit allocation signal supplied from the adaptive bit allocation circuit described later through the terminal 28. The data encoded in this way is
It is taken out through the output terminals 22, 24 and 26. Further, in the adaptive bit allocation encoding circuits 16, 17, and 18, a scale factor indicating what kind of signal magnitude is normalized and a bit length indicating what kind of bit length is quantized. Information is also sought, and these are also output from the output terminals 22, 24 and 26 at the same time.

Next, the specific configuration and operation of the adaptive bit allocation circuit for performing the above bit allocation will be described with reference to FIG. The example of FIG. 7 corresponds to the bit allocation for 6 channels out of the 8 channels corresponding to FIG. That is, it corresponds to 6 channels of a left channel, a left center channel, a center channel, a subwoofer channel, a right center channel and a right channel, which are channels composed of signals that have a great influence on the auditory sense.

In FIG. 7, the common part of each channel will be described by using, for example, the left channel (Lch) (the other channels will be denoted by the same reference numerals and description thereof will be omitted). Input information signal is applied to the input terminal 31 for the left channel.
The terminal 31 corresponds to the terminal 29 of FIG. This input information signal is a mapping circuit (Mapping) 32.
Is expanded from the time domain signal to the frequency domain. Here, in the case of using a filter, time domain samples are obtained as subband signals, and in the case of orthogonal transform output and in the case of performing orthogonal transform after filtering, frequency domain samples are obtained.

These samples are
ing) circuit 33 collects a plurality of samples.
Here, a plurality of samples in the time domain are combined when using a filter, and a plurality of samples in the frequency domain are combined when an orthogonal transform output and when orthogonal transform is performed after filtering.

Further, in this specific example, the time change of the MDCT input time domain signal during the mapping is calculated by the time change calculation circuit 34.

Each sample collected by the blocking circuit 33 into a plurality of samples is normalized by a normalizing circuit 37. Here, the scale factor, which is a coefficient for normalization, is obtained by the scale factor calculation circuit 35. At the same time, the tonality magnitude is calculated by the tonality calculation circuit 36.

The parameters obtained as described above are used by the bit allocation circuit 38 for bit allocation. here,
Assuming that the number of bits that can be used for transmission or recording by expressing MDCT coefficients is 800 Kbps for all channels (the above-mentioned 6 channels), the bit allocation circuit 38 of this specific example uses a first bit allocation (first bit allocation) including channel bit allocation (first bit allocation). Of the channel bit allocation) and a second bit allocation not including the channel bit allocation (second bit allocation amount).

First, the allocation method of the first bit allocation including the channel bit allocation will be described. Here, bit distribution is adaptively performed by observing the distribution of the scale factor in the frequency domain.

In this case, effective bit allocation can be performed by observing the frequency domain distribution of the scale factors of all channels and allocating bits among the channels. At this time, considering that the signal information of multiple channels reaches the left and right ears after being mixed in the same sound field as in the case of a speaker, it is thought that masking works with the sum of all channel signals. Therefore, as shown in A and F of FIG. 8, it is effective to perform bit allocation so that each channel has the same noise level in the same band.
One method for this purpose is to perform bit allocation proportional to the size of the scale factor index. That is, bit allocation is performed by the following formula.

Bm = B * (ΣSFn) / S S = Σ (ΣSFn)

Here, Bm is the bit allocation amount to each channel, B is the bit allocation amount to all channels, and SFn is a scale factor index, which corresponds to the logarithm of the approximate peak value. n is the block floating band number in each channel, m is the channel number, and S is the sum of the scale factor indexes of all channels. In FIG. 8, only the left channel and the right channel are shown and the other four channels are not shown.

In addition to the above, the bit allocation circuit 38 has a process of detecting the time change characteristic of the signal of each channel and changing the bit allocation amount for each channel by this index. The index showing this time change is obtained as follows.

As shown in FIG. 9A to FIG.
Assuming that there is a channel, the bit allocation time block, which is the time unit of bit allocation for the information input signal of each channel, is temporally divided into four, and the peak value of each time block (sub-block) is obtained. Then, bits are shared among channels according to the magnitude of the difference where the peak value of each sub-block changes from small to large. Here, assuming that C bits can be used in a total of 6 channels for this bit allocation, the magnitudes of the differences at which the peak value of each sub-block of each channel changes from small to large are a, b, c, respectively. If d, e, and f decibel (dB), then C * a / T, C * b / T, ..., C, respectively.
* F / T bits (bits) can be allocated. Here, T = a + b + c + d + e + f. The larger the signal information becomes, the larger the bit allocation amount for the channel becomes. In FIG. 9, only the left channel, the left center channel and the right channel are shown and the other three channels are not shown.

Next, the second not including channel bit allocation
A method of allocating the bit will be described. here,
As a second bit allocation method not including channel bit allocation, a bit allocation method consisting of two bit allocations will be described. The second bit allocation corresponds to the bit allocation processing in the adaptive bit allocation encoding circuit in FIG.

These two bit allocations are referred to as bit allocation (1) and bit allocation (2), respectively. In the following bit allocation, the bit rate that can be used in each channel is fixedly determined in advance for each channel. For example, of the 6 channels, 2 channels that play an important part such as voice have 14 channels.
A relatively large bit of 7 kbps is used, 2 kbps is assigned to the subwoofer channel at most, and 100 kbps is assigned to the other channels.

First, the bit amount to be used for the bit allocation (1) is decided. For that purpose, the tonality information and the time change information of the signal information (b) of the spectrum information of the signal information (a) are used.

Tonality information will be described. As an index, a value obtained by dividing the sum of absolute values of differences between adjacent values of the signal spectrum by the number of signal spectra is used as the index. More simply, the average value of the difference between adjacent scale factor indexes of the scale factor for each block for block floating is used. The scale factor index corresponds to the logarithmic value of the rough scale factor. In the present embodiment, the maximum amount of bits to be used for bit allocation (1) is 80, corresponding to the value representing this tonality.
kbps, and the minimum is set to 10 kbps. Here, for simplification, the allocation of all channels is equalized to 10
It is set to 0 kbps.

The tonality calculation is performed by the following equation.

T = (1 / WLmax) (ΣABS (SFn−1))

Note that WLmax is the maximum word length = 16, and SFn is a scale factor index and corresponds to the logarithm of the approximate peak value. n is a block floating band number.

The tonality information T thus obtained is associated with the bit allocation amount of the bit allocation (1) as shown in FIG.

In addition, in this embodiment, the division ratio between the bit allocation (1) and at least one other bit allocation added thereto depends on the time change characteristic of the information signal. In this specific example, the peak value of the signal information is compared for each adjacent block for each time interval obtained by further dividing the orthogonal transform time block size, thereby detecting and increasing the time region in which the amplitude of the information signal rapidly increases. The division rate is determined according to the degree of the state.

The calculation of the time change rate is performed by the following equation.

Vt = ΣVm Vav = (1 / Vmax) * (1 / Ch) Vt

Here, Vt represents the change in peak value of the time sub-block of each channel from small to large in dB value, but Vm is the sum about the channel, and Vm is from small to large peak value of the time sub-block of each channel. The change in dB is expressed in dB and is the largest (however, the maximum value is limited to 30 dB and expressed as Vmax. M is the channel number, Ch is the number of channels, and Vav is the peak value of the time sub-block from small to large. It is the channel average of the change in dB in dB.

The time change rate Va thus obtained
v and the allocation amount of the bit allocation (1) are associated with each other as shown in FIG. Finally, the amount of allocation to bit allocation (1) is calculated by the following formula.

B = 1/2 (Bf + Bt)

Here, B is the amount of allocation to the final bit allocation (1), Bf is the amount of bit allocation obtained from Tva, and B is
t is the bit allocation amount obtained from Vav.

The bit allocation (1) here is allocated in the frequency and time domain depending on the scale factor.

After the bit amount used for the bit allocation (1) is determined in this way, the allocation for the bits not used in the bit allocation (1), that is, the bit allocation (2) is determined next. . Various bit allocations are performed here.

First, a uniform distribution is made for all sample values. FIG. 12 shows an example of the quantization noise spectrum with respect to the bit allocation in this case. In this case, the noise level can be reduced uniformly over the entire frequency band.

Secondly, bit allocation is performed to obtain an auditory effect having dependency on the frequency spectrum and level of signal information. FIG. 13 shows an example of the quantization noise spectrum with respect to the bit allocation in this case. In this example, bit allocation is performed depending on the spectrum of the information signal, and in particular, bit allocation with weighting is performed on the low frequency side of the spectrum of the information signal, and masking on the low frequency side that occurs compared to the wide area side is performed. It compensates for the decrease in effect. This is based on the asymmetry of the masking curve that attaches importance to the low frequency side of the spectrum, considering masking between adjacent critical bands. As described above, in the example of FIG. 13, bit allocation is performed with emphasis on the low frequency band.

Finally, the sum of the bit allocation (1) and the value of the bit allocation added to the bit allocation (1) is taken by the bit allocation circuit 38 of FIG. The final bit allocation is given as the sum of the above bit allocations.

12 and 13, S is the signal spectrum, NL1 is the noise level due to uniform distribution to all the samples, and NL2 is an auditory sound having a positive dependence on the frequency spectrum and level. It shows the noise level due to the bit allocation to get high.

Next, another method of bit allocation not including channel bit allocation will be described below. The operation of the adaptive bit allocation circuit in this case will be described with reference to FIG. 14, the magnitude of the MDCT coefficient is obtained for each block, and the MDCT coefficient is supplied to the input terminal 801. The MDCT coefficient supplied to the input terminal 801 is the energy calculation circuit 8 for each band.
Given to 03. The energy calculation circuit 803 for each band calculates the signal energy for each band obtained by further dividing the critical band or the critical band in the high band. Energy calculation circuit 80 for each band
The energy for each band calculated in 3 is supplied to the energy-dependent bit allocation circuit 804.

In the energy-dependent bit allocation circuit 804,
The available total bit from the available total bit generation circuit 802, which is a ratio of 128 Kbps in this embodiment (100 Kbps in this embodiment), is used to perform bit allocation so as to generate white quantization noise. At this time, the higher the tonality of the input signal, that is, the larger the unevenness of the spectrum of the input signal, the more the bit amount becomes 128K.
The ratio to bps increases. In addition, in order to detect the unevenness of the spectrum of the input signal, the sum of the absolute values of the differences of the block floating coefficients of the adjacent blocks is used as an index. Then, regarding the available bit amount obtained,
Bit allocation is performed in proportion to the logarithmic value of the energy of each band.

The bit allocation calculation circuit 805 which depends on the permissible noise level for hearing first finds the permissible noise amount for each critical band in consideration of the so-called masking effect, etc., based on the spectral data divided for each critical band. Bits obtained by subtracting energy-dependent bits from the total available bits are distributed so as to give a perceptible noise spectrum to the. The energy-dependent bits thus obtained and the bits depending on the permissible noise level of the hearing are added, and the adaptive bit allocation coding circuits 16, 17, and 18 shown in FIG. In each or in the high frequency band, each spectrum data (or MDCT coefficient data) is requantized according to the number of bits assigned to the band obtained by further dividing the critical band into a plurality of bands. The data encoded in this way is output to the output terminals 22, 24, 26 of FIG.
Taken out through.

The auditory permissible noise spectrum calculation circuit in the bit allocation circuit 805 depending on the permissible auditory noise spectrum will be described in more detail. The MDCT circuits 13 and 1 will be described below.
The MDCT coefficients obtained in 4 and 15 are given to the allowable noise calculating circuit.

FIG. 15 is a block circuit diagram showing a schematic configuration of a specific example in which the allowable noise calculating circuits are collectively described. In FIG. 15, the input terminal 521 has an MDC
The spectrum data in the frequency domain is supplied from the T circuits 13, 14, and 15.

The input data in the frequency domain is sent to the energy calculation circuit 522 for each band, and the energy of each critical band (critical band) is calculated, for example, as the sum of squared amplitude values in the band. It is required by doing. Instead of the energy for each band, a peak value, an average value, etc. of the amplitude value may be used. As an output from the energy calculation circuit 522, for example, the spectrum of the total sum value of each band is generally called a Bark spectrum. FIG. 16 shows the Bark spectrum SB for each such critical band. However, in FIG. 16, in order to simplify the illustration, the number of bands of the critical band is represented by 12 bands (B1 to B12).

Here, in order to consider the influence of the so-called masking of the Bark spectrum SB, a convolution process is performed such that the Bark spectrum SB is multiplied by a predetermined weighting function and added. Therefore, the output of the energy calculation circuit 522 for each band, that is, each value of the Bark spectrum SB is sent to the convolution filter circuit 523. The convolution filter circuit 523
Is, for example, a plurality of delay elements that sequentially delay input data, and a plurality of multipliers (for example, 25 multipliers corresponding to each band) that multiply outputs from these delay elements by a filter coefficient (weighting function). , A sum total adder that sums the outputs of the respective multipliers. Note that the masking is a phenomenon in which one signal is masked by another signal and becomes inaudible due to human auditory characteristics.The masking effect includes a time-axis masking effect by an audio signal in the time domain. And there is the same time masking effect by the signal in the frequency domain. Due to these masking effects, even if there is noise in the masked portion, this noise cannot be heard. Therefore, in the actual audio signal, the noise within the masked range is regarded as an 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 and a coefficient of 0.007 by a multiplier M + 3, the convolution processing of the Bark spectrum SB is performed. However, M is 1 to
It is an arbitrary integer of 25.

Next, the output of the convolution filter circuit 523 is sent to the subtractor 524. The subtractor 524 calculates a level α corresponding to an allowable noise level described later in the convoluted area. The level α corresponding to the permissible noise level (permissible noise level) is a level at which the critical noise band becomes the permissible noise level for each band by performing inverse convolution processing, as described later. is there. 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 permissible 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, a> 0, and S is the intensity of the convolution-processed Bark spectrum. In the equation, (n-ai) is the tolerance function. As an example, n = 38 and a = -0.5 can be used.

In this way, the level α is obtained, and this data is transmitted to the divider 526. The divider 526 is for deconvolution of the level α in the convolved area. Therefore, by performing the 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 above-mentioned inverse convolution processing requires complicated calculation, in this embodiment, the inverse convolution is performed using the simplified divider 526.

Next, the masking threshold is transmitted 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 SB, so that the Bark spectrum SB shows the masking threshold MS as shown in FIG.
The level below the level indicated by will be masked. The delay circuit 529 is provided in order to delay the Bark spectrum SB from the energy detection circuit 522 in consideration of the delay amount in each circuit before the synthesis circuit 527.

The output from the subtracter 528 is taken out through the output terminal 531 through the allowable noise correction circuit 530, and, for example, the ROM in which the distribution bit number information is stored in advance.
Etc. (not shown). The ROM or the like is distributed for each band according to the output (the level of the difference between the energy of each band and the output of the noise level setting means) obtained from the subtraction circuit 528 through the allowable noise correction circuit 530. Outputs bit number information.

In this way, the energy-dependent bit and the bit depending on the permissible auditory noise level are added, and the distribution bit number information is supplied to the adaptive bit distribution coding circuits 16, 17, and 18 through the terminal 28 of FIG. By being transmitted, each spectrum data in the frequency domain from the MDCT circuits 13, 14, 15 is quantized by the number of bits assigned to each band.

In summary, in the adaptive bit allocation coding circuits 16, 17, and 18, the energy of each band band (critical band) of the above critical band or the band energy obtained by further dividing the critical band into a plurality of bands in the high band. Alternatively, the spectrum data for each band is quantized by the number of bits distributed according to the level of the difference between the peak value and the output of the noise level setting means.

By the way, at the time of synthesizing by the above-mentioned synthesizing circuit 527, data showing a so-called minimum audible curve RC which is the human auditory characteristic as shown in FIG. 17 supplied from the minimum audible curve generating circuit 532. The masking threshold MS can be combined. In this minimum audible curve, if the absolute noise level is below this minimum audible curve, the noise will not be heard. Even if the coding is the same, this minimum audible curve is different due to the difference in the playback volume during playback,
In a realistic digital system, there is not much difference in how music is put into a 16-bit dynamic range, so if the quantization noise in the most audible frequency band around 4 kHz is inaudible, for example, other It is considered that quantization noise below the level of this minimum audible curve is inaudible in the frequency band. Therefore, for example, the dynamic range of the system is 4 kHz.
Assuming that the noise is not heard in the vicinity, and the allowable noise level is obtained by synthesizing the minimum audible curve RC and the masking threshold MS together, the allowable noise level in this case is shown in FIG. It will be possible to go up to the part shown by the diagonal line of. In addition,
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. 17 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 of the equal loudness curve sent from the correction information output circuit 533, for example. Here, the equal loudness curve is a characteristic curve relating to human auditory characteristics, for example, a curve obtained by obtaining the sound pressure of sound at each frequency that sounds the same as a pure tone of 1 kHz, and connecting the curves. Also called sensitivity curve. Further, this equal loudness curve is the minimum audible curve R shown in FIG.
It draws a curve substantially the same as C. In this equal loudness curve, for example, in the vicinity of 4 kHz, even if the sound pressure is reduced by 8 to 10 dB from 1 kHz, it sounds as loud as 1 kHz. It doesn't sound the same.
Therefore, it is understood that the noise exceeding the level of the minimum audible curve (allowable noise level) should have the frequency characteristic given by the curve corresponding to the equal loudness curve. From this, it can be seen that correcting the permissible noise level in consideration of the equal loudness curve is suitable for human hearing characteristics.

The spectrum shape depending on the permissible noise level of hearing described above is created by bit allocation using a certain ratio of the total usable bits of 128 Kbps. This ratio decreases as the tonality of the input signal increases.

Next, a bit amount dividing method between the two bit allocation methods will be described. Returning to FIG. 14, the MDC
The signal from the input terminal 801 to which the T circuit output is supplied is
The spectrum smoothness calculation circuit 808 is also applied to calculate the spectrum smoothness. In this embodiment, a value obtained by dividing the sum of the absolute values of the differences between the adjacent values of the absolute value of the signal spectrum by the sum of the absolute values of the signal spectrum is calculated as the smoothness of the spectrum.

The above-described spectrum smoothness calculation circuit 808
Is supplied to the bit division rate determination circuit 809, and the bit division rate between the energy-dependent bit allocation and the bit allocation according to the perceptual noise spectrum is determined. The bit division rate is the smoothness calculation circuit 80 of the spectrum.
It is considered that the larger the output value of 8, the smoother the spectrum is, and the bit allocation is performed with more emphasis on the bit allocation by the perceptual noise spectrum than the energy-dependent bit allocation. The bit division ratio determination circuit 809 sends control outputs to multipliers 811 and 812 which control the amount of bit distribution depending on the energy and the permissible noise spectrum of the auditory sense, respectively. Here, if the spectrum is smooth and the output of the bit division rate determination circuit 809 to the multiplier 811 is 0.8 so as to emphasize the energy-dependent bit allocation.
, The output of the bit division rate determination circuit 809 to the multiplier 812 is 1-0.8 = 0.2. The output of these two multipliers is the adder 806.
Are added together to form final bit allocation information, which is output from the output terminal 807.

The state of bit allocation at this time is shown in FIGS. 20 and 21 show the states of quantization noise corresponding to this. FIG. 18 shows a case where the spectrum of the signal is relatively flat, and FIG. 19 shows a case where the signal spectrum shows a high tonality. Also,
In FIG. 18 and FIG. 19, QS indicates the bit amount corresponding to the signal level, and QN in the diagrams indicates the bit amount corresponding to the bit allocation depending on the permissible noise level. 20 and 21, L represents the signal level, NS represents the noise reduction due to the signal level dependency, and NN represents the noise reduction due to the bit allocation that depends on the permissible hearing noise level. .

First, in FIG. 18 showing the case where the spectrum of the signal is flat, the bit allocation depending on the permissible noise level of the hearing aids in obtaining a large signal-to-noise ratio over the entire band. However, relatively low bit allocations are used in the low and high frequencies. This is because auditory sensitivity to noise in this band is low. The bit allocation depending on the signal energy level is small in amount, but in this case, it is concentrated in the high frequency region of the signal level in the middle and low frequencies so as to generate a white noise spectrum.

On the other hand, as shown in FIG. 19, when the signal spectrum exhibits a high tonality, the bit allocation amount depending on the signal energy level increases, and the quantization noise is reduced in a very narrow band. Used to reduce The concentration of the bit allocation depending on the permissible noise level of the auditory sense is less intense.

As shown in FIG. 14, the improvement of the characteristics in the isolated spectrum input signal is achieved by the sum of the bit allocations of the both.

Using the two bit allocations including the channel bit allocations and the bit allocations not including the channel bit allocations obtained as described above, the first and second quantization are performed as follows.

Description will be made with reference to FIG. In this example, among all 6 channels, the channels to which the bit allocation including the channel bit allocation exceeds 147 kbps are the center channel, the subwoofer channel, and the write channel.

First, for a channel in which the bit allocation amount including the channel bit allocation exceeds 147 kbps, it is divided into a certain fixed bit amount, for example, a portion where the maximum is 128 kbps and a portion where it exceeds 128 kbps.

FIG. 23 shows a configuration for performing this processing. Figure 2
In the configuration of No. 3, the normalization process, that is, the block floating, is performed on the block for each of a plurality of samples for each sample of the bit allocation in which the allocation amount in the bit allocation including the channel bit allocation exceeds 147 kbps. At this time, a scale factor is obtained as a coefficient indicating how much block floating has been performed.

In FIG. 23, the MDCT coefficient (MDCT sample) supplied to the input terminal 900 is subjected to normalization processing for each block, that is, block floating, by the normalization circuit 905. At this time, a scale factor is obtained as a coefficient indicating how much block floating has been performed.

Next-stage first quantizer 901
Performs quantization with each sample word length of bit allocation not including the channel bit allocation. At this time, quantization by rounding is performed in order to reduce the quantization noise.

Next, the output of the normalization circuit 905 and the output of the quantizer 901 are sent to the difference unit 902. That is, the difference unit 902 takes the difference (quantization error) between the input and output of the quantizer 901. This differencer 902
Is further sent to the second quantizer 903 via the normalization circuit 906.

In the second quantizer 903, the word length of the difference between each sample word length of the bit allocation including the channel bit allocation and each sample word length of the bit allocation not including the channel bit allocation is determined for each sample. Used for. The floating coefficient at this time is automatically determined from the floating coefficient used in the first quantizer 901 and the word length. That is, when the word length used in the first quantizer 901 is N bits, the floating coefficient used in the second quantizer 903 is obtained at (2 ** N).

Further, in the second quantizer 903, the bit allocation including the rounding processing is performed similarly to the first quantizer 901. In this way, by two quantization,
147 kb in bit allocation including half of the channel bits
The bit of the channel which received the bit allocation exceeding ps is
It is divided into a bit allocation as close to 128 kbps as possible and a remaining bit allocation as low as 128 kbps or less.

Here, 128 kbps and 147 kbps
The reason why two thresholds are provided is as follows. That is, since the remaining bit allocation data also needs sub information indicating the word length, 147 kbps is set as the lower limit amount for bit allocation so that the data area can be obtained including the sub information amount. Also, when the bit allocation amount including the channel bit allocation exceeds 128 kbps and falls below 147 kbps, only sub information can be written in the data portion exceeding 128 kbps, so that there is no room to write sample information, which is meaningless. Therefore, in such a case, in order to perform bit allocation which is smaller than 128 kbps and which is as close as possible to 128 kbps, the channel allocation does not include the channel bit allocation.
8 kbps is set.

Further, a channel whose bit allocation including the channel bit allocation is smaller than 128 kbps uses the bit allocation as it is.

As described above, the magnitudes of the components of the remaining bit allocation are as shown in FIG.
Since the scale factor can be calculated from the scale factor and the word length in (1), only the word length is
Needed by da.

In this way, the quantizers 901 and 903 are
In the above, the rounded and highly efficient quantized output is obtained.

In the structure (decoder) corresponding to the structure (encoder) of FIG. 23, the normalizing circuits 905, 9 are used.
Denormalization circuit 90 for performing denormalization processing corresponding to 06
8 and 907 are provided, and these denormalization circuits 908 and 9 are provided.
The output of 07 is added by the adder 904. The added output is taken out from the output terminal 910.

Next, FIG. 24 shows a specific configuration of the high-rate compression coding circuits 223 and 224 for allocating bits between the surround left channel and the surround right channel shown in FIG.

In FIG. 24, an input terminal 301 is supplied with a surround left channel digital audio signal, and an input terminal 311 is supplied with a surround right channel digital audio signal.

The digital audio signals from the input terminals 301 and 311 are sent to the corresponding buffers 30.
2, 312 once. This buffer 302,3
From No. 12, data is taken out in blocks of N points (N samples) each of which overlaps by 50%.
This block-unit data is used for the orthogonal transformation circuits 303 and 3
13 and MDCT and MDST (Modified Discrete S
ine transform).

The coefficient data from the orthogonal transform circuit 303 is compressed by the corresponding subband block floating point compression circuits 304 and 314, respectively. The coefficient data from the sub-band / block floating point compression circuits 304 and 314 and sub-information such as word length information and scale factor are sent to the corresponding adaptive quantization circuits 305 and 315. Also, the sub-band block floating point compression circuits 304, 314
From, the spectral information is sent to the log spectral envelope detection circuit 308.

The adaptive quantization circuit 305 determines the coefficient data based on the bit allocation information from the distribution determination circuit 309 which determines the inter-channel bit allocation amount based on the envelope information detected by the log spectral envelope detection circuit 308. And sub information is adaptively quantized. The adaptive quantization circuits 305 and 315 output quantized coefficient data, sub information, and bit allocation information. The outputs of the adaptive quantization circuits 305 and 315 are the same as the multiplex error correction circuit 3 described above.
It is sent to 06, 316.

The multiplex error correction circuits 306 and 316 multiplex the quantized coefficient data, sub information and bit allocation information for each channel, and add an error correction code. These multiplex error correction circuits 306 and 31
The outputs from the output terminals 6 and 6 via the output terminals 307 and 317 are the outputs of the high-rate compression encoding circuits 223 and 224 for the surround left channel and the surround right channel of FIG.

Next, FIG. 25 shows a configuration of a high-audibility decompression / decoding circuit corresponding to each of the high-audibility compression / encoding circuits 217 to 222 shown in FIG. That is, the high-audibility expansion decoding circuit of FIG. 25 is applied to the receiving device corresponding to the transmitting device of the present invention, and the above-mentioned m channels among the n-channel digital audio signals are used. Is a circuit (for one channel) that decodes a signal that has been compression-encoded by the first encoding method for the digital audio signal.

In FIG. 25, the quantized MDCT coefficients of each band are the decoding device input terminals 122 and 124,
The block size information and adaptive bit allocation information given to and used by 126 are input terminals 123, 125, 1
Given to 27. Decoding circuits 116, 117, 118
Then, the adaptive bit allocation information is used to cancel the bit allocation, and the block size information is used to perform decompression decoding.

Next, the IMDCT circuits 113, 114, 1
At 15, the frequency domain signal is transformed into a time domain signal. IQMF circuits 112 and 111 decode the time domain signals of these subbands into whole-domain signals.

Here, in the high-audibility decompression decoding circuit, a channel for which bit allocation (1) of 128 kbps or less including the channel bit allocation is performed and a bit allocation (2) of 147 kbps or more including the channel bit allocation are performed. A certain amount of bits in the channel to be addressed, eg 128k
The decoding circuit 116, 117, 118 has a maximum bps portion and a portion exceeding 128 kbps, respectively.
Is decrypted with. However, the two parts of the bit allocation (2) become high-precision samples by adding the respective samples after they are decoded.

Regarding the way of arranging the obtained data of each channel, in the sync block, first, (1) a channel in which the bit allocation of 128 kbps or less including the channel bit allocation is performed, and (2) the channel bit allocation Including a certain bit amount in a channel in which a bit allocation of 147 kbps or more is performed, for example, 128 kbp
The part having the maximum s is arranged in the order of channels, and then the part exceeding 128 kbps in the channel in which the bit allocation of 147 kbps or more including the channel bit allocation is performed is arranged in the channel order.

Next, the high-rate compression coding circuit 223 of FIG.
FIG. 26 shows the configuration of a high rate expansion decoding circuit corresponding to H.224.
(1 channel). That is, the high-rate decompression decoding circuit of FIG. 26 is applied to the receiving device corresponding to the transmitting device of the present invention, and the second above-mentioned second is applied to the digital audio signals of the nm channels. This is a configuration of a circuit (for one channel) that decodes a signal that has been compression-encoded by an encoding method.

In FIG. 26, the input terminal 410 is supplied with the high-rate compression-encoded digital audio signal. This signal is demultiplexed
The error correction circuit 411 performs demultiplexing and error correction.

The demultiplex error correction circuit 411 outputs adaptively quantized coefficient data, sub information, and bit allocation information. The coefficient data and the sub information are sent to the adaptive inverse quantization circuit 412.
Also, the bit allocation information is sent to the quantization step size control circuit 413. The adaptive inverse quantization circuit 41
2 is the quantization step size control circuit 41
Inverse quantization is performed on the quantized transform coefficient information based on the quantization step size information from 3. The quantized compression transform coefficient from the adaptive dequantization circuit 412 is sent to the subband block floating point expansion circuit 414.

In the sub-band / block floating-point expansion circuit 414, the sub-band / block floating-point compression circuits 304 and 3 shown in FIG.
The reverse process of 14 is performed. The output of the decompression circuit 414 is converted into N-point sample data by the inverse orthogonal transform circuit 415 which similarly performs the inverse transform processing of the orthogonal transform circuits 303 and 313 of FIG. 24, and is sent to the window overlap addition circuit 416. The window overlap adding circuit 416 removes the overlap and outputs the PCM audio signal. This P
The CM audio signal is taken out from the output terminal 416.

[0135]

As is apparent from the above description, in the digital audio signal transmitting apparatus of the present invention,
Among the signals of a plurality of channels, a signal of a channel consisting of a signal having a high influence on the auditory perception is compression-encoded by the first encoding method with the first compression rate that causes less degradation on the auditory sense, and The second compression rate higher than the first compression rate is applied to the signal of the channel including the low-impact signal.
Since the compression encoding is performed with the encoding method of
High-quality compression coding can be performed for particularly important sounds, and waste of the bit allocation amount (byte allocation amount) can be eliminated.

Further, in the digital audio signal receiving apparatus of the present invention, the first compression rate, which is less deteriorated in hearing than the signal of the channel consisting of the signals having a high auditory influence among the signals of the plurality of channels. Second code having a second compression rate higher than the first compression rate with respect to a channel signal composed of a signal that has been compression-encoded by the first encoding method and a signal that has a low auditory influence. It becomes possible to decode the respective signals that have been compression-encoded by the encoding method.

[Brief description of drawings]

FIG. 1 is a block circuit diagram showing a schematic configuration of a main part of a digital audio signal transmission apparatus according to an embodiment of the present invention.

FIG. 2 is a diagram for explaining the arrangement of speakers in an 8-channel digital surround system.

FIG. 3 is a block circuit diagram showing a configuration for multiplexing a compressed audio signal of each channel.

FIG. 4 is a block circuit diagram showing a schematic configuration of a specific example (an example in which bit allocation between channels is not performed) of a high-audibility compression encoding circuit.

FIG. 5 is a block circuit diagram showing a schematic configuration of a specific example (an example of performing bit allocation between channels) of a high-audibility compression encoding circuit.

FIG. 6 is a diagram showing frequency and time division of a signal in a high-audibility compression encoding circuit.

FIG. 7 is a block circuit diagram showing an example of a configuration for obtaining a parameter for bit allocation in multi-channel of a high-audibility compression encoding circuit.

[Fig. 8] Fig. 8 is a diagram illustrating the concept of bit allocation based on the magnitude of a spectrum among channels in a high-audibility compression encoding circuit.

[Fig. 9] Fig. 9 is a diagram illustrating a method of obtaining a parameter for bit allocation in consideration of a time characteristic of an information signal between channels in a high-audibility compression encoding circuit.

FIG. 10 is a diagram showing the relationship between the bit allocation amount and the tonality of bit allocation (1).

FIG. 11 is a diagram showing a relationship between a bit allocation amount of bit allocation (1) and a time change rate.

FIG. 12 is a diagram showing a noise spectrum at the time of uniform distribution.

FIG. 13 is a diagram showing an example of a noise spectrum due to bit allocation for obtaining an auditory effect having dependency on a frequency spectrum and a level of an information signal.

FIG. 14 is a block circuit diagram showing a configuration for realizing a bit allocation method using the two of the magnitude of an information signal and the permissible noise spectrum of hearing.

FIG. 15 is a block circuit diagram showing a configuration for obtaining an allowable noise level.

FIG. 16 is a diagram showing an example of a masking threshold depending on the signal level of each band.

FIG. 17 is a diagram showing an information spectrum, a masking threshold, and a minimum audible limit.

FIG. 18 is a diagram showing signal level-dependent and auditory permissible noise level-dependent bit allocation for an information signal having low tonality.

FIG. 19 is a diagram showing signal level-dependent and auditory permissible noise level-dependent bit allocation for an information signal having high tonality.

FIG. 20 is a diagram showing a quantization noise level for an information signal having low tonality.

FIG. 21 is a diagram showing a quantization noise level for an information signal having high tonality.

[Fig. 22] Fig. 22 is a diagram illustrating the relationship of bit allocation in multiple channels in the high-hearingness compression encoding circuit.

FIG. 23 is a block circuit diagram showing a specific configuration for dividing bit allocation.

FIG. 24 is a block circuit diagram showing a specific configuration example of a high-rate compression encoding circuit.

FIG. 25 is a block circuit diagram showing a configuration example of a high-audibility extension decoding circuit of the receiving device according to the embodiment of the present invention.

FIG. 26 is a block circuit diagram showing a configuration example of a high-rate expansion decoding circuit of the receiving apparatus according to the embodiment of the present invention.

[Explanation of symbols]

209 to 216 ... A / D converter 217 to 222 ... High-audibility compression encoding circuit 223, 224 ... High-rate compression encoding circuit 225, 226 ... High-pass filter 228, 229 ... Low-pass Filters 227, 230 ... Adders 241 ... Multiplexers 302, 313 ... Buffers 303, 313 ... Orthogonal transformation circuits 304, 314 ... Subband block floating point compression circuits 305, 315 ... Adaptation Quantization circuit 306, 316 ... Multiplex error correction circuit 308 ... Log spectral envelope detection circuit 309 ... Distribution decision circuit 11, 12 ... Band division filter 13, 14, 15 ... MDCT circuit 16, 17, 18 ... Adaptive bit allocation encoding circuit 19, 20, 2・ ・ ・ Block size determination circuit 31 ・ ・ ・ Each channel information signal input terminal 32 ・ ・ ・ Mapping circuit 33 ・ ・ ・ Blocking circuit 34 ・ ・ ・ Time change calculation circuit 35 ・ ・ ・ Scale factor calculation circuit 36 ・ ・ ・Tonality calculation circuit 37 ... Normalization circuit 38 ... Bit allocation circuit 116, 117, 118 ... Adaptive bit allocation decoding circuit 113, 114, 115 ... IMDCT circuit 112, 111 ... IQMF circuit 411 Demultiplex error correction circuit 412 Adaptive dequantization circuit 413 Quantization step size control circuit 414 Subband block floating point expansion circuit 415 Inverse orthogonal transformation circuit 416 ..Window overlap adding circuit 520 ... Allowable noise Calculation circuit 521 ... Allowable noise calculation circuit input terminal 522 ... Energy detection circuit for each band 523 ... Convolution filter circuit 524 ... Subtractor 525 ... n-ai function generation circuit 526 ... -Divider 527 ... Synthesis circuit 528 ... Subtractor 530 ... Allowable noise correction circuit 532 ... Minimum audible curve generation circuit 533 ... Correction information output circuit 802 ... Usable total bit generation Circuit 803 ... Energy calculation circuit for each band 804 ... Energy-dependent bit allocation circuit 805 ... Auditory permissible noise level-dependent bit allocation circuit 806 ... Adder 808 ... Spectral smoothness calculation circuit 809 ... Bit division rate determination circuits 811, 812 ... Multipliers 905, 906 ... Normalization circuit 901 ... First quantizer 90 3 ... Second quantizer 907, 909 ... Inverse normalization circuit 904 ... Adder

[Procedure amendment]

[Submission date] April 8, 1994

[Procedure Amendment 1]

[Document name to be amended] Statement

[Correction target item name] Full text

[Correction method] Change

[Correction content]

[Document name] Statement

Title: Digital audio signal transmitting apparatus and receiving apparatus, and digital audio signal transmitting method and receiving method

[Claims]

Detailed Description of the Invention

[0001]

BACKGROUND OF THE INVENTION 1. Field of the Invention The present invention, for example, compresses and transmits a multi-channel digital audio signal used in a stereo such as a movie film projection system, a video tape recorder, a video disc player, or a so-called multi surround sound system. The present invention relates to a digital audio signal transmitting apparatus and a corresponding receiving apparatus, a digital audio signal transmitting method, and a corresponding receiving method.

[0002]

2. Description of the Related Art Not only in the case of ordinary audio equipment, but also in stereo or multi-surround 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. It is becoming increasingly common to handle multiple channels of audio or voice signals.

In particular, for business use, digital audio has become multi-channel, and, for example, equipment that handles 8-channel digital audio signals has become widespread. An example of a device that handles the 8-channel digital audio signal is a movie film projection system. Also, in stereo or multi-surround sound systems such as high-definition televisions, video tape recorders, video disc players, etc.
A plurality of channels of audio or voice signals such as 8 channels are being handled.

In the motion picture film projection system that handles the 8-channel digital audio signals, for the motion picture film, for example, a left channel, a left center channel, a center channel, a right center channel, a right channel, a surround left channel, a surround right. Recording of 8-channel digital audio signals of channels and subwoofer channels is being performed. It should be noted that the above 8 recorded on the movie film
Each channel is, for example, a left speaker, a left center speaker, a center speaker, a right center speaker, a right speaker, a subwoofer speaker, which is arranged on the screen side on which an image reproduced from the image recording area of the movie film is projected by the projector. , A surround left speaker arranged on the left side and a surround right speaker arranged on the right side so as to surround the auditorium.

[0005]

However, in the case of recording the above 8-channel digital audio signals on a motion picture film, the motion picture film has a sampling frequency of 44. Since it is difficult to secure an area in which 16-bit linearly quantized audio data of 1 kHz can be recorded for 8 channels, it is necessary to compress and record the 8 channels of audio data.

Further, since a medium called a film is apt to have scratches on its surface, if digital data is recorded as it is, the data will be seriously lost and it will not be put to practical use. For this reason, the capability of the error correction code becomes very important, and the data compression must be performed to the extent that it can be recorded in the recording area on the film including the correction code.

However, when compression coding is performed, the musical instrument, human voice, or the like changes from the original sound. Therefore, when it is adopted as a recording format of a medium in which faithful reproduction of the original sound is required as in the above-mentioned movie film. For important sounds such as human voice, some means for improving the sound quality is required.

In view of the above, the present invention enables high-quality compression coding for particularly important sounds and also enables high-compression compression coding. It is an object of the present invention to provide a digital audio signal transmission apparatus using high-efficiency coding and a receiving apparatus corresponding to the transmission apparatus.

[0009]

The present invention has been proposed in order to achieve the above object, and a digital audio signal transmitting apparatus of the present invention is provided with a plurality of channels (n channels, n is greater than 3). Is a transmission device that compresses and encodes a digital audio signal of
Among the digital audio signals of the above n channels, for the digital audio signals of m (n> m) channels and having a higher audible influence than the digital audio signals of the other nm channels. The first
Second compression coding means for compressing and coding by the first coding method having a compression rate of 1) and a second compression rate higher than the first compression rate for the digital audio signals of the n-m channels. Second compression encoding means for performing compression encoding with a second encoding method having a compression rate of, and a compression encoding output from the first compression encoding means and the second compression encoding. It is characterized in that the compression encoded output from the means is transmitted together.

In the transmission device of the present invention, the n channels are a center channel, a left channel, a right channel, a surround left channel and a surround right channel, and the n-m channels are surround left channels. Surround light channel. Alternatively, the n channels are a center channel, a left channel, a right channel, a left center channel, a right center channel, a surround left channel and a surround right channel, and the nm channels are a surround left channel and a surround right channel. Is.

Further, in the transmission apparatus of the present invention, the first encoding method divides the input digital audio signal into a plurality of bands, divides the digital audio signal in each band into blocks for a plurality of samples, and sets each block unit. Is a system for adaptively compressing and coding the spectrum component orthogonally transformed in accordance with the auditory characteristic, and the second coding system is related to coefficient information obtained by orthogonally transforming an input digital audio signal for every plural samples and its relation. Get sub information to
This is a method in which the bit allocation amount of each channel is determined according to the energy of each channel, and compression coding is adaptively performed with this bit allocation amount. The above transmission includes recording on a recording medium.

Further, the digital audio signal receiving apparatus of the present invention is a digital audio signal of a plurality of channels (n channels, n is a positive integer greater than 3),
The first code having the first compression ratio for a digital audio signal of m (n> m) channels, which has a higher auditory influence than other digital audio signals of nm channels. The second encoding method has a second compression rate higher than the first compression rate with respect to the digital audio signals of the n−m channels while being compression-encoded by the encoding method. A receiver for a digital audio signal for receiving a compression-encoded signal, wherein the first encoding method is applied to the digital audio signals of the m channels among the digital audio signals of the n channels. Corresponding to the second encoding method for the digital audio signals of the n−m channels and the first decoding means for performing the first decompression / decoding. It is characterized in that a second decoding means for performing a second decompression and decoding.

Next, a method of transmitting a digital audio signal according to the present invention is a method of transmitting a digital audio signal in which digital audio signals of a plurality of channels (n channels, n is a positive integer greater than 3) are compression-encoded and transmitted. Of the n channels of digital audio signals, m (n> m) channels of digital audio signals are more audibly influential than the other n−m channels of digital audio signals. A first compression coding step of compressing and coding a signal using a first coding method with a first compression ratio, and the first compression coding step for the digital audio signals of the n−m channels. A second compression encoding step of performing compression encoding with a second encoding method having a second compression rate higher than the compression rate, and from the first compression encoding step It is characterized in transmitting the compressed encoded output from the compression encoded output and the second compression encoding step together.

The method of receiving a digital audio signal according to the present invention is, among digital audio signals of a plurality of channels (n channels, n is a positive integer greater than 3),
The first code having the first compression ratio for a digital audio signal of m (n> m) channels, which has a higher auditory influence than other digital audio signals of nm channels. The second encoding method has a second compression rate higher than the first compression rate with respect to the digital audio signals of the n−m channels while being compression-encoded by the encoding method. A method of receiving a digital audio signal for receiving a signal that has been compression-encoded, wherein the first encoding system is applied to the digital audio signals of the m channels among the digital audio signals of the n channels. Corresponding to the second encoding method for the digital audio signals of the n−m channels and the first decoding step of performing the first decompression decoding corresponding to It is characterized in that a second decoding step of performing a second decompression and decoding.

[0015]

According to the apparatus and method for transmitting a digital audio signal of the present invention, of the signals of a plurality of channels, a signal of a channel consisting of a signal having a great influence on the auditory sense is less deteriorated in the auditory sense. A second code having a second compression rate higher than the first compression rate is applied to a signal of a channel composed of a signal having a low auditory influence, by performing compression encoding by the first encoding method with a compression rate. The compression coding is performed by the encoding method.

Further, according to the digital audio signal receiving apparatus and method of the present invention, of the signals of a plurality of channels, the first signal is less deteriorated in the auditory sense than the signal of the channel consisting of the signal having a high auditory influence. Of the second compression rate higher than the first compression rate with respect to the signal of the channel composed of the signal compression-encoded by the first encoding method of the The signals that have been compression-encoded by the encoding method of No. 2 are each decoded.

[0017]

Embodiments of the present invention will be described below with reference to the drawings.

FIG. 1 shows the configuration of a digital audio signal transmission apparatus of the present invention for implementing the digital audio signal transmission method of the present invention. A digital audio signal transmission apparatus according to an embodiment of the present invention is a transmission apparatus that compresses and encodes digital audio signals of n (n is a positive integer greater than 3; for example, 8 in this embodiment) channels. Is.

Here, in this embodiment, the above n pieces (8 pieces) are used.
The left channel (L
ch), left center channel (LCch), center channel (Cch), subwoofer channel (SWc)
h), right center channel (RCch), right channel (Rch), surround left channel (LBc)
h), a surround right channel (RBch) is used. In the present invention, the n channels are not limited to the above example, but may be, for example, 5 channels of a center channel, a left channel, a right channel, a surround left channel and a surround right channel, or a center channel, a left channel and a right channel. It is also possible to use seven channels including a channel, a left center channel, a right center channel, a surround left channel and a surround right channel.

The digital audio signal transmitting apparatus according to the present embodiment is a digital audio signal of m (n> m, m = 6 in the example of FIG. 1) channels of the above eight channels of digital audio signals. Other nm
(In the example of FIG. 1, mn = 8−6 = 2) For a signal that has a higher auditory influence than a digital audio signal of the number of channels, a compression code is generated by the first encoding method with the first compression ratio. The high-audibility compression coding circuits 217 to 222 as the first compression coding means for converting the digital audio signals into the n-m channels and the second compression higher than the first compression ratio. A high-rate compression coding circuit 223, 224 which is a second compression coding means for performing compression coding by a second coding method having a rate.
It is characterized in that the compression coded outputs from 17 to 222 and the compression coded outputs from the high rate compression coding circuits 223 and 224 are both transmitted.

In the present embodiment, among the eight channels, the channels (six channels) made up of the signals that have a high perceptual influence on the signals and are compressed by the first coding method are used as the left channel and the left center. A channel, a center channel, a subwoofer channel, a right center channel and a right channel, and channels (nm channels) compressed by the second encoding method are a surround left channel and a surround right channel. Of course, the present invention is not limited to these, and for example, when the above n channels are, for example, 5 channels of a center channel, a left channel, a right channel, a surround left channel and a surround right channel, the m channels are replaced by the center channel. The left channel and the right channel may be three channels, and the n−m channels may be the surround left channel and the surround right channel. Similarly, when the n channels are, for example, 7 channels of a center channel, a left channel, a right channel, a left center channel, a right center channel, a surround left channel and a surround right channel, the m channels are referred to as center channels. A left channel and a right channel may be used, and the above n−m channels may be a surround left channel and a surround right channel.

Further, in the transmission apparatus of the embodiment of the present invention, the above-described sub-band coding or the like is used for the high-audibility compression coding circuits 217 to 222 which perform compression coding by the first coding method. It is a high-efficiency compression encoding method for audio signals, which makes use of human auditory characteristics,
A method of compressing audio data to about ⅕, that is, a spectrum in which an input digital audio signal is divided into a plurality of bands, the digital audio signal of each band is divided into a plurality of samples, and orthogonally transformed in each block. A method of adaptively compression-coding data according to human auditory characteristics (for example, so-called ATRAC: Adap
tive TRansform Acoustic Coding method) is used.

Further, in the high-rate compression coding circuits 223 and 224 which carry out compression coding by the second coding method, coefficient data obtained by orthogonally transforming an input digital audio signal for every plural samples and sub-information related thereto ( The word length information and the scale factor information) are obtained, the bit allocation of each channel is determined according to the energy of each channel, and the compression encoding is adaptively performed by the bit allocation of each channel. Needless to say, the present invention is not limited to each of these encoding methods.

Further, in the present embodiment, the transmission is as follows.
For example, recording on motion picture film, recording on disk-shaped recording media such as optical disks, magneto-optical disks, phase-change optical disks, magnetic disks, tape-shaped recording media such as magnetic tape, recording on semiconductor memory, IC cards, etc. Can be mentioned.

When the transmission is recorded on a movie film, it corresponds to a digital surround system in which speakers are arranged as shown in FIG. 2, for example. Each channel is a center (C) channel, a subwoofer (SW) channel, a left (L) channel, a left center (CL) channel, a right (R) channel, a right center (CR) channel, a left surround (LB) channel.
There are eight channels, a light surround (RB) channel.

In FIG. 2, each of the eight channels recorded on the motion picture film has a screen 101 on which an image reproduced from an image recording area of the motion picture film is projected by a projector (projector 100). Left speaker 106, left center speaker 104, center speaker 10 arranged on the side
2, light center speaker 105, light speaker 1
07, surround left speakers 108 and 200, surround right speakers 109 and 201, and subwoofer speaker 103.

The center speaker 102 is arranged in the center of the screen 101 side and outputs a reproduced sound based on audio data of the center channel, and outputs the most important reproduced sound such as an actor's dialogue. The subwoofer speaker 103 outputs a reproduced sound based on audio data of the subwoofer channel, and effectively outputs a sound that is felt as vibration rather than a low frequency sound such as an explosion sound, and an explosion scene. Is often used effectively. The left speaker 106 and the right speaker 107 are arranged on the left and right sides of the screen 101, and output a reproduced sound by the left channel audio data and a reproduced sound by the right channel audio data, and exhibit a stereo sound effect. The left center speaker 104 and the right center speaker 105 are arranged between the center speaker 102 and the left speaker 106 and the right speaker 107, and are reproduced by the left center channel audio data and reproduced by the right center channel audio data. It outputs sound and plays an auxiliary role for the left speaker 106 and the right speaker 107, respectively. Particularly in a movie theater or the like where the screen 101 is large and the number of persons accommodated is large, the localization of the sound image tends to become unstable depending on the position of the seat.
4 and the light center speaker 107 are added, it is effective in creating a more realistic localization of the sound image. Further, the surround left speaker 108 and the surround right speaker 109 are arranged so as to surround the spectators' seats, and output a reproduced sound by the audio data of the surround left channel and a reproduced sound by the audio data of the surround right channel. It has the effect of giving the impression of being wrapped in clapping and cheering. As a result, a more stereoscopic sound image can be created.

Returning to FIG. 1, the transmission apparatus of FIG. 1 which handles the 8-channel digital audio signal as described above,
In order to obtain the above 8-channel digital audio signals, the microphones 201 to 208 corresponding to the respective channels are obtained.
Corresponding analog audio signals from A /
The D converters 209 to 216 convert the digital audio signals.

The digital audio signals of the respective channels from the respective A / D converters 209 to 216 are sent to the corresponding compression encoding circuits 217 to 224 and compressed and encoded respectively. Note that each of these compression encoding circuits 217-2
The specific configuration of 24 will be described later.

The above-mentioned high-audibility compression encoding circuits 217 to 222.
The compressed and encoded audio data from are sent to the subsequent stage from the corresponding output terminals 233 to 238.

On the other hand, the high-rate compression coding circuits 223, 2
The compression-encoded audio data from 24 is added by the adder 227 after only the high frequency components are extracted by the high pass filters 225 and 226, respectively. The compression-encoded audio data from the high-rate compression encoding circuits 223 and 224 are also sent to low-pass filters 228 and 229, respectively. The data from the adder 227 is supplied to the low pass filter 2 corresponding to the surround left channel side by the adder 230.
28 output is added and output from the output terminal 231.
The output of the low pass filter 229 corresponding to the surround right channel is output from the output terminal 232.

Here, in the surround left channel and the surround right channel, the outputs from the high pass filters 225 and 226 are added by the adder 227, and the added output of the adder 227 is further added.
The reason for adding to the output of the low-pass filter 228 of the surround left channel by 8 is as follows.

That is, the human ear has less sense of localization for high-frequency components. Therefore, for high-frequency components, no matter which speaker is selected from among a plurality of speakers, human beings recognize which speaker. It is difficult to hear if it is coming out. Therefore, each high frequency component of the audio signals of a plurality of channels is
Even if you send only to the speaker corresponding to the channel,
It is not perceived by humans that the high frequency component of the audio signal of each channel is output from only the one speaker. Therefore, in this embodiment, as described above,
Of the channels, for example, add the high frequency components of the surround left channel and surround right channel audio,
For example, it is added to the sound of the low frequency component of the surround left channel. As a result, the high frequency components of the two channels, the surround left channel and the surround right channel, can be compressed into one channel.
The crosstalk processing between channels in such a high frequency component is performed not only between the surround left channel and the surround right channel as in the example of FIG. 1 but also between other channels (not shown) (for example, The same can be done for all channels). In this way, it is possible to further increase the compression rate by performing channel crosstalk processing of high frequency components for all channels.

The output terminals 233 to 238 shown in FIG.
, 231 and 232 of each channel are sent to the multiplexer 241 via the corresponding terminals 233 to 238 and 231, 232 of FIG. 3, respectively, and are multiplexed. The output of the multiplexer 241 is transmitted to the transmission line via the output terminal 242 or recorded on the above-mentioned movie film of the present invention, the disc-shaped recording medium, the tape-shaped recording medium, or the like.

Next, the high-hearing-compression compression encoding circuits 217 to 222 shown in FIG. 1 will be specifically described.

First, FIG. 4 shows a configuration of a specific example of each of the above-described high-audibility compression encoding circuits 217 to 222.

In the compression coding circuit shown in FIG. 4, the input digital signal is divided into a plurality of frequency bands by a filter and the orthogonal transformation is performed for each frequency band to obtain the spectrum data of the frequency axis. The bits are adaptively allocated and encoded for each so-called critical band (critical band) in consideration of human auditory characteristics described later. At this time, a band obtained by further dividing the critical bandwidth is used in the high band. Of course, the non-blocking frequency division width by a filter or the like may be an equal division width. Further, in this embodiment, the block size (block length) is adaptively changed according to the input signal before the orthogonal transformation, and the critical bandwidth (critical band) is further subdivided in the critical band unit or in the high range. Floating process is performed in the block.

As the band division filter,
For example, there is a filter such as QMF, which is 1976 RECro
chiere Digital coding of speech in subbands Bell
Syst.Tech. J. Vol.55, No.8 1976. Also ICASSP 83, BOSTONPolyphase Quadrature f
ilters-A new subband coding technique Joseph H.Ro
Thweiler describes an equal bandwidth filter partitioning method and apparatus.

Further, the critical band is a frequency band divided in consideration of human auditory characteristics,
It is a band of a certain pure tone when the pure tone is masked by a narrow band noise having the same strength near the frequency of the pure tone. This critical band has a wider bandwidth as it goes higher, and the entire frequency band of 0 to 22 kHz is divided into, for example, 25 critical bands.

That is, in FIG. 4, an audio PCM signal of 0 to 22 kHz, for example, is supplied to the input terminal 10. This input signal is divided into 0 to 11 kHz band and 1 by a band division filter 11 such as so-called QMF.
The signal of 0 to 11 kHz band is divided into the 1 to 22 kHz band and the 0 to 5.5 kHz band and 5.5 to 11 kHz by the band dividing filter 12 such as so-called QMF.
and z-band. 1 from the band division filter 11
A signal in the 1 kHz to 22 kHz band is an MDCT (Modified Discrete Cosine Transform) that is an example of an orthogonal transformation circuit.
5.5 sent from the band division filter 12 to the circuit 13.
A signal in the k to 11 kHz band is sent to the MDCT circuit 14, and a signal in the 0 to 5.5 kHz band from the band division filter 12 is sent to the MDCT circuit 15 to be MDCT processed.

In each of the MDCT circuits 13, 14 and 15, MDCT processing is performed based on the block size determined by the block determining circuits 19, 20 and 21 provided for each band as described later.

The MDCT processing in each MDCT circuit 13, 14, 15 is described in ICASSP 1987 Sub.
band / Transform Coding Using Filter Bank Designs B
asedon Time Domain Aliasing Cancellation JPPrinc
en ABBradley Univ. of Surrey Royal Melbourne Ins
t.of Tech.

The information indicating the block size decided by the block decision circuits 19, 20, 21 is sent to adaptive bit allocation coding circuits 16, 17, 18 which will be described later, and output from the output terminals 23, 25, 27. To be done.

Here, the adaptive bit allocation encoding circuit 1
6, 17 and 18, the MDCT circuits 13 and 1
From the outputs of 4 and 15, the energy of each critical band (critical band) or a band obtained by further dividing the critical band in the high band is calculated, for example, the root mean square of each amplitude value in the band is calculated. Sought by
Bit allocation is performed based on the calculation result. Of course, the scale factor itself may be used for subsequent bit allocation. In this case, the calculation of new energy is not required, and the hardware scale is saved. Further, instead of the energy for each band, it is also possible to use the peak value, the average value, etc. of the amplitude values. The spectrum data in the frequency domain or the MDCT coefficient data obtained by the MDCT processing in each MDCT circuit 13, 14, 15 is a so-called critical band (critical band) or a band in which a critical band is further divided in a high band. The data are collected and sent to the adaptive bit allocation coding circuits 16, 17, and 18.

Further, the adaptive bit allocation encoding circuit 1
6, 17, and 18 each spectrum data (or MDCT) according to the block size information and the number of bits assigned to each band obtained by further dividing the critical band (critical band) or the critical band in the high band.
The coefficient data) is requantized (normalized and quantized). Adaptive bit allocation coding circuits 16, 17, 1
The data encoded in 8 are output terminals 22, 24, 26
Taken out through. Further, in the adaptive bit allocation encoding circuits 16, 17, and 18, a scale factor indicating what kind of signal magnitude is normalized and a bit length indicating what kind of bit length is quantized. Information is also sought, and these are also output terminals 22,
It is output from 24 and 26.

The data from the output terminals 22 to 27 are put together and the high-audibility compression encoding circuits 217 to 2 are combined.
22 output.

In the examples of FIGS. 1 and 4, the high-audibility compression coding circuits 217 to 222 allocate bits to each channel to perform compression coding. It is also possible to perform bit allocation among the encoding circuits 217 to 222 (that is, perform bit allocation among channels corresponding to the circuits 217 to 222).

The high-audibility compression encoding circuits 217 to 222
Each circuit configuration in the case of performing bit allocation between channels will be described below. FIG. 5 shows the configuration of a high-audibility compression encoding circuit that allocates bits among channels. In the configuration of FIG. 5, adaptive bit allocation coding circuits 16, 17, 1
Other components except 8 are basically the same as the corresponding components in FIG.

In the compression encoding circuit shown in FIG. 5, a concrete example of the block size in each MDCT circuit 13, 14, 15 determined by the block determination circuits 19, 20, 21 similar to that in FIG. 4 is shown in FIG. Shown in A and B. 6A shows a case where the orthogonal transform block size is long (orthogonal transform block size in the long mode), FIG.
Shows the case where the orthogonal transform block size is short (orthogonal transform block size in the short mode).
In the specific example of FIG. 6, the three filter outputs are
Each has two orthogonal transform block sizes. That is, for a signal in the low band of 0 to 5.5 kHz and a signal in the mid band of 5.5 to 11 kHz, the number of samples in one block is set to a long block length (A in FIG. 6). When 128 blocks are selected and a short block is selected (B in FIG. 6), the number of samples in one block is a block for every 32 samples. On the other hand, the high side 1
For a signal in the 1 kHz to 22 kHz band, if the block length is long (A in FIG. 6), the number of samples in one block is 2
56 samples and a short block is selected (Fig. 6
In B), the number of samples in one block is a block for every 32 samples. When a short block is selected in this way, the number of samples of orthogonal transform blocks in each band is set to be the same, the time resolution is increased in the higher frequency range, and the number of windows used for blocking is reduced. In addition, the block determination circuits 19, 20, and 21 of the specific example of FIG.
The information indicating the block size determined in step 1 is sent to the adaptive bit allocation coding circuits 16, 17 and 18, which will be described later, and is also output from the output terminals 23, 25 and 27.

In the adaptive bit allocation coding circuits 16, 17, and 18 of the specific example of FIG. 5, the above block size information and the critical band (critical band) or in the high frequency band, the critical band is further divided and allocated. Re-quantization (normalized and quantized) of each spectrum data (or MDCT coefficient data) according to the number of bits
I am trying to do it. At this time, the adaptive bit allocation coding circuits 16, 17, and 18 adaptively and optimally allocate the used bit amount for each channel by observing the channel bit allocation among the channels, that is, the entire signal of each channel. Allocate simultaneously. The channel bit allocation in this case is performed based on the channel bit allocation signal supplied from the adaptive bit allocation circuit described later through the terminal 28. The data encoded in this way is
It is taken out through the output terminals 22, 24 and 26. Further, in the adaptive bit allocation encoding circuits 16, 17, and 18, a scale factor indicating what kind of signal magnitude is normalized and a bit length indicating what kind of bit length is quantized. Information is also sought, and these are also output from the output terminals 22, 24 and 26 at the same time.

Next, the specific configuration and operation of the adaptive bit allocation circuit for performing the above bit allocation will be described with reference to FIG. The example of FIG. 7 corresponds to the bit allocation for 6 channels out of the 8 channels corresponding to FIG. That is, it corresponds to 6 channels of a left channel, a left center channel, a center channel, a subwoofer channel, a right center channel and a right channel, which are channels composed of signals that have a great influence on the auditory sense.

In FIG. 7, the common part of each channel will be described using, for example, a left channel (Lch) (the other channels are denoted by the same reference numerals and the description thereof will be omitted). Input information signal is applied to the input terminal 31 for the left channel.
The terminal 31 corresponds to the terminal 29 of FIG. This input information signal is a mapping circuit (Mapping) 32.
Is expanded from the time domain signal to the frequency domain. Here, in the case of using a filter, time domain samples are obtained as subband signals, and in the case of orthogonal transform output and in the case of performing orthogonal transform after filtering, frequency domain samples are obtained.

These samples are
ing) circuit 33 collects a plurality of samples.
Here, a plurality of samples in the time domain are combined when using a filter, and a plurality of samples in the frequency domain are combined when an orthogonal transform output and when orthogonal transform is performed after filtering.

Further, in this example, the time change calculation circuit 34 calculates the time change of the MDCT input time domain signal during the mapping.

Each sample collected by the blocking circuit 33 into a plurality of samples is normalized by a normalizing circuit 37. Here, the scale factor, which is a coefficient for normalization, is obtained by the scale factor calculation circuit 35. At the same time, the tonality magnitude is calculated by the tonality calculation circuit 36.

The parameters calculated as described above are used by the bit allocation circuit 38 for bit allocation. here,
Assuming that the number of bits that can be used for transmission or recording by expressing MDCT coefficients is 800 Kbps for all channels (the above-mentioned 6 channels), the bit allocation circuit 38 of this specific example uses a first bit allocation (first bit allocation) including channel bit allocation (first bit allocation). Of the channel bit allocation) and a second bit allocation not including the channel bit allocation (second bit allocation amount).

First, the allocation method of the first bit allocation including the channel bit allocation will be described. Here, bit distribution is adaptively performed by observing the distribution of the scale factor in the frequency domain.

In this case, effective bit allocation can be performed by observing the distribution of the scale factors of all channels in the frequency domain and allocating bits among the channels. At this time, considering that the signal information of multiple channels reaches the left and right ears after being mixed in the same sound field as in the case of a speaker, it is thought that masking works with the sum of all channel signals. Therefore, as shown in A and F of FIG. 8, it is effective to perform bit allocation so that each channel has the same noise level in the same band.
One method for this purpose is to perform bit allocation proportional to the size of the scale factor index. That is, bit allocation is performed by the following formula.

Bm = B * (ΣSFn) / S S = Σ (ΣSFn)

Here, Bm is a bit allocation amount to each channel, B is a bit allocation amount to all channels, SFn is a scale factor index, and corresponds to the logarithm of the approximate peak value. n is the block floating band number in each channel, m is the channel number, and S is the sum of the scale factor indexes of all channels. In FIG. 8, only the left channel and the right channel are shown and the other four channels are not shown.

In addition to the above, the bit allocation circuit 38 has a process of detecting the time change characteristic of the signal of each channel and changing the bit allocation amount for each channel by this index. The index showing this time change is obtained as follows.

As shown in FIG. 9A to FIG.
Assuming that there is a channel, the bit allocation time block, which is the time unit of bit allocation for the information input signal of each channel, is temporally divided into four, and the peak value of each time block (sub-block) is obtained. Then, bits are shared among channels according to the magnitude of the difference where the peak value of each sub-block changes from small to large. Here, assuming that C bits can be used in a total of 6 channels for this bit allocation, the magnitudes of the differences at which the peak value of each sub-block of each channel changes from small to large are a, b, c, respectively. If d, e, and f decibel (dB), then C * a / T, C * b / T, ..., C, respectively.
* F / T bits (bits) can be allocated. Here, T = a + b + c + d + e + f. The larger the signal information becomes, the larger the bit allocation amount for the channel becomes. In FIG. 9, only the left channel, the left center channel and the right channel are shown and the other three channels are not shown.

Next, the second not including channel bit allocation
A method of allocating the bit will be described. here,
As a second bit allocation method not including channel bit allocation, a bit allocation method consisting of two bit allocations will be described. The second bit allocation corresponds to the bit allocation processing in the adaptive bit allocation encoding circuit in FIG.

These two bit allocations are referred to as bit allocation (1) and bit allocation (2), respectively. In the following bit allocation, the bit rate that can be used in each channel is fixedly determined in advance for each channel. For example, of the 6 channels, 2 channels that play an important part such as voice have 14 channels.
A relatively large bit of 7 kbps is used, 2 kbps is assigned to the subwoofer channel at most, and 100 kbps is assigned to the other channels.

First, the bit amount to be used for the bit allocation (1) is decided. For that purpose, the tonality information and the time change information of the signal information (b) of the spectrum information of the signal information (a) are used.

Here, the tonality information will be described. As the index, a value obtained by dividing the sum of absolute values of differences between adjacent values of the signal spectrum by the number of signal spectra is used as the index. More simply, the average value of the difference between adjacent scale factor indexes of the scale factor for each block for block floating is used. The scale factor index corresponds to the logarithmic value of the rough scale factor. In the present embodiment, the maximum amount of bits to be used for bit allocation (1) is 80, corresponding to the value representing this tonality.
kbps, and the minimum is set to 10 kbps. Here, for simplification, the allocation of all channels is equalized to 10
It is set to 0 kbps.

The tonality calculation is performed by the following equation.

T = (1 / WLmax) (ΣABS (SFn−1))

WLmax is a word length maximum value = 16, and SFn is a scale factor index and corresponds to the logarithm of the approximate peak value. n is a block floating band number.

The tonality information T thus obtained is associated with the bit allocation amount of the bit allocation (1) as shown in FIG.

In addition to this, in the present embodiment, the division ratio between the bit allocation (1) and at least one other bit allocation added thereto depends on the time change characteristic of the information signal. In this specific example, the peak value of the signal information is compared for each adjacent block for each time interval obtained by further dividing the orthogonal transform time block size, thereby detecting and increasing the time region in which the amplitude of the information signal rapidly increases. The division rate is determined according to the degree of the state.

The time change rate calculation is performed by the following equation.

Vt = ΣVm Vav = (1 / Vmax) * (1 / Ch) Vt

Here, Vt is the sum of the values of the change of the peak value of the time subblock of each channel from small to large expressed in dB, and Vm is the sum of the peak value of the time subblock of each channel from small to large. Change in dB value, the size of the largest one (however, the maximum value is 30d
It is limited to B and is represented by Vmax. m is the channel number, C
h is the number of channels, and Vav is the channel average of the change in peak value of the time subblock from small to large expressed in dB.

The time change rate Va thus obtained
v and the allocation amount of the bit allocation (1) are associated with each other as shown in FIG. Finally, the amount of allocation to bit allocation (1) is calculated by the following formula.

B = 1/2 (Bf + Bt)

Here, B is the final bit allocation amount (1), Bf is the bit allocation amount obtained from Tva, and B is
t is the bit allocation amount obtained from Vav.

The bit allocation (1) here is allocated in the frequency and time domains depending on the scale factor.

When the bit amount used for the bit allocation (1) is determined in this way, the allocation for the bits not used in the bit allocation (1), that is, the bit allocation (2) is determined next. . Various bit allocations are performed here.

First, a uniform distribution is made for all sample values. FIG. 12 shows an example of the quantization noise spectrum with respect to the bit allocation in this case. In this case, the noise level can be reduced uniformly over the entire frequency band.

Secondly, bit allocation is performed to obtain an auditory effect having dependency on the frequency spectrum and level of signal information. FIG. 13 shows an example of the quantization noise spectrum with respect to the bit allocation in this case. In this example, bit allocation is performed depending on the spectrum of the information signal, and in particular, bit allocation with weighting is performed on the low frequency side of the spectrum of the information signal, and masking on the low frequency side that occurs compared to the wide area side is performed. It compensates for the decrease in effect. This is based on the asymmetry of the masking curve that attaches importance to the low frequency side of the spectrum, considering masking between adjacent critical bands. As described above, in the example of FIG. 13, bit allocation is performed with emphasis on the low frequency band.

Finally, the sum of the bit allocation (1) and the value of the bit allocation added to the bit allocation (1) is taken by the bit allocation circuit 38 of FIG. The final bit allocation is given as the sum of the above bit allocations.

12 and 13, S is the signal spectrum, NL1 is the noise level due to uniform distribution for all the samples, and NL2 is the auditory sense with a positive dependence on the frequency spectrum and level. It shows the noise level due to the bit allocation to get high.

Next, another method of bit allocation not including channel bit allocation will be described below. The operation of the adaptive bit allocation circuit in this case will be described with reference to FIG. 14, the magnitude of the MDCT coefficient is obtained for each block, and the MDCT coefficient is supplied to the input terminal 801. The MDCT coefficient supplied to the input terminal 801 is the energy calculation circuit 8 for each band.
Given to 03. The energy calculation circuit 803 for each band calculates the signal energy for each band obtained by further dividing the critical band or the critical band in the high band. Energy calculation circuit 80 for each band
The energy for each band calculated in 3 is supplied to the energy-dependent bit allocation circuit 804.

In the energy-dependent bit allocation circuit 804,
The available total bit from the available total bit generation circuit 802, which is a ratio of 128 Kbps in this embodiment (100 Kbps in this embodiment), is used to perform bit allocation so as to generate white quantization noise. At this time, the higher the tonality of the input signal, that is, the larger the unevenness of the spectrum of the input signal, the more the bit amount becomes 128K.
The ratio to bps increases. In addition, in order to detect the unevenness of the spectrum of the input signal, the sum of the absolute values of the differences of the block floating coefficients of the adjacent blocks is used as an index. Then, regarding the available bit amount obtained,
Bit allocation is performed in proportion to the logarithmic value of the energy of each band.

The bit allocation calculation circuit 805, which depends on the permissible noise level for hearing, first determines the permissible noise amount for each critical band in consideration of the so-called masking effect, etc., based on the spectral data divided for each critical band. Bits obtained by subtracting energy-dependent bits from the total available bits are distributed so as to give a perceptible noise spectrum to the. The energy-dependent bits thus obtained and the bits depending on the permissible noise level of the hearing are added, and the adaptive bit allocation coding circuits 16, 17, and 18 shown in FIG. In each or in the high frequency band, each spectrum data (or MDCT coefficient data) is requantized according to the number of bits assigned to the band obtained by further dividing the critical band into a plurality of bands. The data encoded in this way is output to the output terminals 22, 24, 26 of FIG.
Taken out through.

The auditory permissible noise spectrum calculation circuit in the bit allocation circuit 805 depending on the permissible auditory noise spectrum will be described in more detail.
The MDCT coefficients obtained in 4 and 15 are given to the allowable noise calculating circuit.

FIG. 15 is a block circuit diagram showing a schematic configuration of a specific example in which the allowable noise calculating circuits are collectively described. In FIG. 15, the input terminal 521 has an MDC
The spectrum data in the frequency domain is supplied from the T circuits 13, 14, and 15.

The input data in the frequency domain is sent to the energy calculation circuit 522 for each band, and the energy of each critical band (critical band) is calculated, for example, as the sum of squared amplitude values in the band. It is required by doing. Instead of the energy for each band, a peak value, an average value, etc. of the amplitude value may be used. As an output from the energy calculation circuit 522, for example, the spectrum of the total sum value of each band is generally called a Bark spectrum. FIG. 16 shows the Bark spectrum SB for each such critical band. However, in FIG. 16, in order to simplify the illustration, the number of bands of the critical band is represented by 12 bands (B1 to B12).

Here, in order to consider the influence of so-called masking of the Bark spectrum SB, a convolution process is performed such that the Bark spectrum SB is multiplied by a predetermined weighting function and added. Therefore, the output of the energy calculation circuit 522 for each band, that is, each value of the Bark spectrum SB is sent to the convolution filter circuit 523. The convolution filter circuit 523
Is, for example, a plurality of delay elements that sequentially delay input data, and a plurality of multipliers (for example, 25 multipliers corresponding to each band) that multiply outputs from these delay elements by a filter coefficient (weighting function). , A sum total adder that sums the outputs of the respective multipliers. Note that the masking is a phenomenon in which one signal is masked by another signal and becomes inaudible due to human auditory characteristics.The masking effect includes a time-axis masking effect by an audio signal in the time domain. And there is the same time masking effect by the signal in the frequency domain. Due to these masking effects, even if there is noise in the masked portion, this noise cannot be heard. Therefore, in the actual audio signal, the noise within the masked range is regarded as an 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 and a coefficient of 0.007 by a multiplier M + 3, the convolution processing of the Bark spectrum SB is performed. However, M is 1 to
It is an arbitrary integer of 25.

Next, the output of the convolution filter circuit 523 is sent to the subtractor 524. The subtractor 524 calculates a level α corresponding to an allowable noise level described later in the convoluted area. The level α corresponding to the permissible noise level (permissible noise level) is a level at which the critical noise band becomes the permissible noise level for each band by performing inverse convolution processing, as described later. is there. 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 permissible 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, a> 0, and S is the intensity of the convolution-processed Bark spectrum. In the equation, (n-ai) is the tolerance function. As an example, n = 38 and a = -0.5 can be used.

In this way, the level α is obtained, and this data is transmitted to the divider 526. The divider 526 is for deconvolution of the level α in the convolved area. Therefore, by performing the 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 above-mentioned inverse convolution processing requires complicated calculation, in this embodiment, the inverse convolution is performed using the simplified divider 526.

Next, the masking threshold is transmitted 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 SB, so that the Bark spectrum SB shows the masking threshold MS as shown in FIG.
The level below the level indicated by will be masked. The delay circuit 529 is provided in order to delay the Bark spectrum SB from the energy detection circuit 522 in consideration of the delay amount in each circuit before the synthesis circuit 527.

The output from the subtracter 528 is taken out via the output terminal 531 via the allowable noise correction circuit 530, and, for example, the ROM in which the distribution bit number information is stored in advance.
Etc. (not shown). The ROM or the like is distributed for each band according to the output (the level of the difference between the energy of each band and the output of the noise level setting means) obtained from the subtraction circuit 528 through the allowable noise correction circuit 530. Outputs bit number information.

In this way, the energy-dependent bits and the bits depending on the permissible hearing noise level are added, and the distributed bit number information is supplied to the adaptive bit allocation coding circuits 16, 17, and 18 via the terminal 28 of FIG. By being transmitted, each spectrum data in the frequency domain from the MDCT circuits 13, 14, 15 is quantized by the number of bits assigned to each band.

In summary, in the adaptive bit allocation coding circuits 16, 17, and 18, the energy of each critical band of the critical band or the energy of a band obtained by further dividing the critical band into a plurality of bands in the high band. Alternatively, the spectrum data for each band is quantized by the number of bits distributed according to the level of the difference between the peak value and the output of the noise level setting means.

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 a human auditory characteristic as shown in FIG. The masking threshold MS can be combined. In this minimum audible curve, if the absolute noise level is below this minimum audible curve, the noise will not be heard. Even if the coding is the same, this minimum audible curve is different due to the difference in the playback volume during playback,
In a realistic digital system, there is not much difference in how music is put into a 16-bit dynamic range, so if the quantization noise in the most audible frequency band around 4 kHz is inaudible, for example, other It is considered that quantization noise below the level of this minimum audible curve is inaudible in the frequency band. Therefore, for example, the dynamic range of the system is 4 kHz.
Assuming that the noise is not heard in the vicinity, and the allowable noise level is obtained by synthesizing the minimum audible curve RC and the masking threshold MS together, the allowable noise level in this case is shown in FIG. It will be possible to go up to the part shown by the diagonal line of. In addition,
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. 17 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 of the equal loudness curve sent from the correction information output circuit 533, for example. Here, the equal loudness curve is a characteristic curve relating to human auditory characteristics, for example, a curve obtained by obtaining the sound pressure of sound at each frequency that sounds the same as a pure tone of 1 kHz, and connecting the curves. Also called sensitivity curve. Further, this equal loudness curve is the minimum audible curve R shown in FIG.
It draws a curve substantially the same as C. In this equal loudness curve, for example, in the vicinity of 4 kHz, even if the sound pressure is reduced by 8 to 10 dB from 1 kHz, it sounds as loud as 1 kHz. It doesn't sound the same.
Therefore, it is understood that the noise exceeding the level of the minimum audible curve (allowable noise level) should have the frequency characteristic given by the curve corresponding to the equal loudness curve. From this, it can be seen that correcting the permissible noise level in consideration of the equal loudness curve is suitable for human hearing characteristics.

The spectrum shape depending on the permissible hearing noise level described above is created by bit allocation using a certain ratio of the total usable bits of 128 Kbps. This ratio decreases as the tonality of the input signal increases.

Next, a bit amount dividing method between the two bit allocation methods will be described. Returning to FIG. 14, the MDC
The signal from the input terminal 801 to which the T circuit output is supplied is
The spectrum smoothness calculation circuit 808 is also applied to calculate the spectrum smoothness. In this embodiment, a value obtained by dividing the sum of the absolute values of the differences between the adjacent values of the absolute value of the signal spectrum by the sum of the absolute values of the signal spectrum is calculated as the smoothness of the spectrum.

The spectrum smoothness calculation circuit 808 described above.
Is supplied to the bit division rate determination circuit 809, and the bit division rate between the energy-dependent bit allocation and the bit allocation according to the perceptual noise spectrum is determined. The bit division rate is the smoothness calculation circuit 80 of the spectrum.
It is considered that the larger the output value of 8, the smoother the spectrum is, and the bit allocation is performed with more emphasis on the bit allocation by the perceptual noise spectrum than the energy-dependent bit allocation. The bit division ratio determination circuit 809 sends control outputs to multipliers 811 and 812 which control the amount of bit distribution depending on the energy and the permissible noise spectrum of the auditory sense, respectively. Here, if the spectrum is smooth and the output of the bit division rate determination circuit 809 to the multiplier 811 is 0.8 so as to emphasize the energy-dependent bit allocation.
, The output of the bit division rate determination circuit 809 to the multiplier 812 is 1-0.8 = 0.2. The output of these two multipliers is the adder 806.
Are added together to form final bit allocation information, which is output from the output terminal 807.

The state of bit allocation at this time is shown in FIGS. 20 and 21 show the states of quantization noise corresponding to this. FIG. 18 shows a case where the spectrum of the signal is relatively flat, and FIG. 19 shows a case where the signal spectrum shows a high tonality. Also,
In FIG. 18 and FIG. 19, QS indicates the bit amount corresponding to the signal level, and QN in the diagrams indicates the bit amount corresponding to the bit allocation depending on the permissible noise level. 20 and 21, L represents the signal level, NS represents the noise reduction due to the signal level dependency, and NN represents the noise reduction due to the bit allocation that depends on the permissible hearing noise level. .

First, in FIG. 18 showing the case where the spectrum of the signal is flat, the bit allocation depending on the permissible hearing noise level is useful for obtaining a large signal to noise ratio over the entire band. However, relatively low bit allocations are used in the low and high frequencies. This is because auditory sensitivity to noise in this band is low. The bit allocation depending on the signal energy level is small in amount, but in this case, it is concentrated in the high frequency region of the signal level in the middle and low frequencies so as to generate a white noise spectrum.

On the other hand, as shown in FIG. 19, when the signal spectrum shows a high tonality, the bit allocation amount depending on the signal energy level becomes large, and the quantization noise is reduced in a very narrow band. Used to reduce The concentration of the bit allocation depending on the permissible noise level of the auditory sense is less intense.

As shown in FIG. 14, the improvement of the characteristics in the isolated spectrum input signal is achieved by the sum of the bit allocations of the both.

Using the two bit allocations including the channel bit allocations and the bit allocations not including the channel bit allocations obtained as described above, the first and second quantization are performed as follows.

Description will be made with reference to FIG. In this example, among all 6 channels, the channels to which the bit allocation including the channel bit allocation exceeds 147 kbps are the center channel, the subwoofer channel, and the write channel.

First, for a channel in which the bit allocation amount including the channel bit allocation exceeds 147 kbps, it is divided into a certain fixed bit amount, for example, a portion where the maximum is 128 kbps and a portion where it exceeds 128 kbps.

FIG. 23 shows a configuration for performing this processing. Figure 2
In the configuration of No. 3, the normalization process, that is, the block floating, is performed on the block for each of a plurality of samples for each sample of the bit allocation in which the allocation amount in the bit allocation including the channel bit allocation exceeds 147 kbps. At this time, a scale factor is obtained as a coefficient indicating how much block floating has been performed.

In FIG. 23, the MDCT coefficient (MDCT sample) supplied to the input terminal 900 is subjected to a normalization process for each block, that is, a block floating process by the normalization circuit 905 every plural samples. At this time, a scale factor is obtained as a coefficient indicating how much block floating has been performed.

Next-stage first quantizer 901
Performs quantization with each sample word length of bit allocation not including the channel bit allocation. At this time, quantization by rounding is performed in order to reduce the quantization noise.

Next, the output of the normalization circuit 905 and the output of the quantizer 901 are sent to the difference unit 902. That is, the difference unit 902 takes the difference (quantization error) between the input and output of the quantizer 901. This differencer 902
Is further sent to the second quantizer 903 via the normalization circuit 906.

In the second quantizer 903, the word length of the difference between each sample word length of the bit allocation including the channel bit allocation and each sample word length of the bit allocation not including the channel bit allocation is determined for each sample. Used for. The floating coefficient at this time is automatically determined from the floating coefficient used in the first quantizer 901 and the word length. That is, when the word length used in the first quantizer 901 is N bits, the floating coefficient used in the second quantizer 903 is obtained at (2 ** N).

In the second quantizer 903, bit allocation including rounding processing is performed in the same manner as the first quantizer 901. In this way, by two quantization,
147 kb in bit allocation including half of the channel bits
The bit of the channel which received the bit allocation exceeding ps is
It is divided into a bit allocation as close to 128 kbps as possible and a remaining bit allocation as low as 128 kbps or less.

Here, 128 kbps and 147 kbps
The reason why two thresholds are provided is as follows. That is, since the remaining bit allocation data also needs sub information indicating the word length, 147 kbps is set as the lower limit amount for bit allocation so that the data area can be obtained including the sub information amount. Also, when the bit allocation amount including the channel bit allocation exceeds 128 kbps and falls below 147 kbps, only sub information can be written in the data portion exceeding 128 kbps, so that there is no room to write sample information, which is meaningless. Therefore, in such a case, in order to perform bit allocation which is smaller than 128 kbps and which is as close as possible to 128 kbps, the channel allocation does not include the channel bit allocation.
8 kbps is set.

Further, a channel whose bit allocation including the channel bit allocation is smaller than 128 kbps uses the bit allocation as it is.

As described above, the magnitudes of the components of the remaining bit allocation are as shown in FIG.
Since the scale factor can be calculated from the scale factor and the word length in (1), only the word length is
Needed by da.

In this way, the quantizers 901 and 903 are
In the above, the rounded and highly efficient quantized output is obtained.

In the structure (decoder) corresponding to the structure (encoder) of FIG. 23, the above normalizing circuits 905, 9 are used.
Denormalization circuit 90 for performing denormalization processing corresponding to 06
8 and 907 are provided, and these denormalization circuits 908 and 9 are provided.
The output of 07 is added by the adder 904. The added output is taken out from the output terminal 910.

Next, FIG. 24 shows a specific configuration of the high-rate compression coding circuits 223 and 224 for allocating bits between the surround left channel and the surround right channel shown in FIG.

In FIG. 24, an input terminal 301 is supplied with a surround left channel digital audio signal, and an input terminal 311 is supplied with a surround right channel digital audio signal.

Digital audio signals from the input terminals 301 and 311 are transferred to the corresponding buffers 30.
2, 312 once. This buffer 302,3
From No. 12, data is taken out in blocks of N points (N samples) each of which overlaps by 50%.
This block-unit data is used for the orthogonal transformation circuits 303 and 3
13 and MDCT and MDST (Modified Discrete S
ine transform).

The coefficient data from the orthogonal transform circuit 303 is compressed by the corresponding subband block floating point compression circuits 304 and 314, respectively. The coefficient data from the sub-band / block floating point compression circuits 304 and 314 and sub-information such as word length information and scale factor are sent to the corresponding adaptive quantization circuits 305 and 315. Also, the sub-band block floating point compression circuits 304, 314
From, the spectral information is sent to the log spectral envelope detection circuit 308.

The adaptive quantization circuit 305 determines the coefficient data based on the bit allocation information from the distribution determination circuit 309 which determines the inter-channel bit allocation amount based on the envelope information detected by the log spectral envelope detection circuit 308. And sub information is adaptively quantized. The adaptive quantization circuits 305 and 315 output quantized coefficient data, sub information, and bit allocation information. The outputs of the adaptive quantization circuits 305 and 315 are the same as the multiplex error correction circuit 3 described above.
It is sent to 06, 316.

These multiplex error correction circuits 306 and 316 multiplex the quantized coefficient data, sub information and bit allocation information for each channel, and add an error correction code. These multiplex error correction circuits 306 and 31
The outputs from the output terminals 6 and 6 via the output terminals 307 and 317 are the outputs of the high-rate compression encoding circuits 223 and 224 for the surround left channel and the surround right channel of FIG.

Next, FIG. 25 shows the configuration of a high-audibility decompression / decoding circuit corresponding to each of the high-audibility compression / encoding circuits 217 to 222 shown in FIG. That is, the high-audibility expansion decoding circuit of FIG. 25 is applied to the receiving apparatus and method corresponding to the transmitting apparatus and method of the present invention, and among the n plural-channel digital audio signals, This is a configuration of a circuit (for one channel) that decodes a signal that is compression-encoded by the first encoding method with respect to digital audio signals of m channels.

In FIG. 25, the quantized MDCT coefficients of each band are the decoding device input terminals 122 and 124,
The block size information and adaptive bit allocation information given to and used by 126 are input terminals 123, 125, 1
Given to 27. Decoding circuits 116, 117, 118
Then, the adaptive bit allocation information is used to cancel the bit allocation, and the block size information is used to perform decompression decoding.

Next, the IMDCT circuits 113, 114, 1
At 15, the frequency domain signal is transformed into a time domain signal. IQMF circuits 112 and 111 decode the time domain signals of these subbands into whole-domain signals.

Here, in the high-audibility decompression decoding circuit, a channel for which bit allocation (1) of 128 kbps or less including the channel bit allocation is performed and a bit allocation (2) of 147 kbps or more including the channel bit allocation are performed. A certain amount of bits in the channel to be addressed, eg 128k
The decoding circuit 116, 117, 118 has a maximum bps portion and a portion exceeding 128 kbps, respectively.
Is decrypted with. However, the two parts of the bit allocation (2) become high-precision samples by adding the respective samples after they are decoded.

Regarding the way of arranging the obtained data of each channel, in the sync block, first, (1) the channel in which the bit allocation of 128 kbps or less including the channel bit allocation is performed, and (2) the channel bit allocation Including a certain bit amount in a channel in which a bit allocation of 147 kbps or more is performed, for example, 128 kbp
The part having the maximum s is arranged in the order of channels, and then the part exceeding 128 kbps in the channel in which the bit allocation of 147 kbps or more including the channel bit allocation is performed is arranged in the channel order.

Next, the high-rate compression coding circuit 223 of FIG.
FIG. 26 shows the configuration of a high rate expansion decoding circuit corresponding to H.224.
(1 channel). That is, the high-rate decompression decoding circuit of FIG. 26 is applied to the receiving device corresponding to the transmitting device of the present invention, and the second above-mentioned second is applied to the digital audio signals of the nm channels. This is a configuration of a circuit (for one channel) that decodes a signal that has been compression-encoded by an encoding method.

In FIG. 26, the input terminal 410 is supplied with the high-rate compression-encoded digital audio signal. This signal is demultiplexed
The error correction circuit 411 performs demultiplexing and error correction.

The demultiplex error correction circuit 411 outputs the adaptively quantized coefficient data, sub information, and bit allocation information. The coefficient data and the sub information are sent to the adaptive inverse quantization circuit 412.
Also, the bit allocation information is sent to the quantization step size control circuit 413. The adaptive inverse quantization circuit 41
2 is the quantization step size control circuit 41
Inverse quantization is performed on the quantized transform coefficient information based on the quantization step size information from 3. The quantized compression transform coefficient from the adaptive dequantization circuit 412 is sent to the subband block floating point expansion circuit 414.

In the sub-band / block floating-point expansion circuit 414, the sub-band / block floating-point compression circuits 304 and 3 shown in FIG.
The reverse process of 14 is performed. The output of the decompression circuit 414 is converted into N-point sample data by the inverse orthogonal transform circuit 415 which similarly performs the inverse transform processing of the orthogonal transform circuits 303 and 313 of FIG. 24, and is sent to the window overlap addition circuit 416. The window overlap adding circuit 416 removes the overlap and outputs the PCM audio signal. This P
The CM audio signal is taken out from the output terminal 416.

[0137]

As is clear from the above description, in the digital audio signal transmitting apparatus and method of the present invention, among the signals of a plurality of channels, a signal of a channel consisting of a signal having a high auditory influence is applied. For example, the compression coding is performed by the first coding method with the first compression rate that causes less deterioration in the auditory sense, and the compression rate is lower than that of the first compression rate for the signal of the channel including the signal that has a low auditory influence. Since the compression encoding is performed by the second encoding method having the high second compression rate, it is possible to perform the high-quality sound compression encoding for the particularly important sound, and the bit allocation amount (byte It is possible to eliminate the waste of the allocation amount.

Further, in the digital audio signal receiving apparatus and method of the present invention, of the signals of a plurality of channels, there is less deterioration in the auditory sense with respect to the signal of the channel consisting of the signal having a high auditory influence. A second compression rate higher than the first compression rate with respect to a channel signal composed of a signal compression-encoded by the first compression rate encoding method and a signal having a low auditory influence. It becomes possible to respectively decode the signals that have been compression-encoded by the encoding method of.

[Brief description of drawings]

FIG. 1 is a block circuit diagram showing a schematic configuration of a main part of a digital audio signal transmission apparatus according to an embodiment of the present invention.

FIG. 2 is a diagram for explaining the arrangement of speakers in an 8-channel digital surround system.

FIG. 3 is a block circuit diagram showing a configuration for multiplexing a compressed audio signal of each channel.

FIG. 4 is a block circuit diagram showing a schematic configuration of a specific example (an example in which bit allocation between channels is not performed) of a high-audibility compression encoding circuit.

FIG. 5 is a block circuit diagram showing a schematic configuration of a specific example (an example of performing bit allocation between channels) of a high-audibility compression encoding circuit.

FIG. 6 is a diagram showing frequency and time division of a signal in a high-audibility compression encoding circuit.

FIG. 7 is a block circuit diagram showing an example of a configuration for obtaining a parameter for bit allocation in multi-channel of a high-audibility compression encoding circuit.

[Fig. 8] Fig. 8 is a diagram illustrating the concept of bit allocation based on the magnitude of a spectrum among channels in a high-audibility compression encoding circuit.

[Fig. 9] Fig. 9 is a diagram illustrating a method of obtaining a parameter for bit allocation in consideration of a time characteristic of an information signal between channels in a high-audibility compression encoding circuit.

FIG. 10 is a diagram showing the relationship between the bit allocation amount and the tonality of bit allocation (1).

FIG. 11 is a diagram showing a relationship between a bit allocation amount of bit allocation (1) and a time change rate.

FIG. 12 is a diagram showing a noise spectrum at the time of uniform distribution.

FIG. 13 is a diagram showing an example of a noise spectrum due to bit allocation for obtaining an auditory effect having dependency on a frequency spectrum and a level of an information signal.

FIG. 14 is a block circuit diagram showing a configuration for realizing a bit allocation method using the two of the magnitude of an information signal and the permissible noise spectrum of hearing.

FIG. 15 is a block circuit diagram showing a configuration for obtaining an allowable noise level.

FIG. 16 is a diagram showing an example of a masking threshold depending on the signal level of each band.

FIG. 17 is a diagram showing an information spectrum, a masking threshold, and a minimum audible limit.

FIG. 18 is a diagram showing signal level-dependent and auditory permissible noise level-dependent bit allocation for an information signal having low tonality.

FIG. 19 is a diagram showing signal level-dependent and auditory permissible noise level-dependent bit allocation for an information signal having high tonality.

FIG. 20 is a diagram showing a quantization noise level for an information signal having low tonality.

FIG. 21 is a diagram showing a quantization noise level for an information signal having high tonality.

[Fig. 22] Fig. 22 is a diagram illustrating the relationship of bit allocation in multiple channels in the high-hearingness compression encoding circuit.

FIG. 23 is a block circuit diagram showing a specific configuration for dividing bit allocation.

FIG. 24 is a block circuit diagram showing a specific configuration example of a high-rate compression encoding circuit.

FIG. 25 is a block circuit diagram showing a configuration example of a high-audibility extension decoding circuit of the receiving device according to the embodiment of the present invention.

FIG. 26 is a block circuit diagram showing a configuration example of a high-rate expansion decoding circuit of the receiving apparatus according to the embodiment of the present invention.

[Description of Codes] 209 to 216 ... A / D converters 217 to 222 ... High-audibility compression coding circuit 223, 224 ... High-rate compression coding circuit 225, 226 ... High-pass filter 228, 229 ... Low-pass filter 227, 230 ... Adder 241 ... Multiplexer 302, 313 ... Buffer 303, 313 ... Orthogonal transformation circuit 304, 314 ... Subband block floating point compression circuit 305, 315 ... Adaptive quantization circuit 306, 316 ... Multiplex error correction circuit 308 ... Log spectral envelope detection circuit 309 ... Distribution decision circuit 11, 12 ... Band division filter 13, 14, 15 ... MDCT circuit 16, 17, 18 ... Adaptive bit allocation encoding circuit 9, 20, 21 ... Block size determination circuit 31 ... Each channel information signal input terminal 32 ... Mapping circuit 33 ... Blocking circuit 34 ... Time change calculation circuit 35 ... Scale factor calculation Circuit 36 ... Tonality calculation circuit 37 ... Normalization circuit 38 ... Bit allocation circuit 116, 117, 118 ... Adaptive bit allocation decoding circuit 113, 114, 115 ... IMDCT circuit 112, 111. ..IQMF circuit 411 ... Demultiplex error correction circuit 412 ... Adaptive inverse quantization circuit 413 ... Quantization step size control circuit 414 ... Subband block floating point expansion circuit 415 ... Inverse Orthogonal transformation circuit 416 ... Window overlap addition circuit 52 ... Allowable noise calculation circuit 521 ... Allowable noise calculation circuit input terminal 522 ... Energy detection circuit for each band 523 ... Convolution filter circuit 524 ... Subtractor 525 ... n-ai function Generation circuit 526 ... Divider 527 ... Synthesis circuit 528 ... Subtractor 530 ... Allowable noise correction circuit 532 ... Minimum audible curve generation circuit 533 ... Correction information output circuit 802 ... Total usable bit generation circuit 803 ... Energy calculation circuit for each band 804 ... Energy dependent bit allocation circuit 805 ... Auditory permissible noise level dependent bit allocation circuit 806 ... Adder 808 ... Spectrum Smoothness calculation circuit 809 ... Bit division rate determination circuit 811, 812 ... Multiplier 905, 906 ... Normalization circuit 901 ... First Quantizer 903 ... Second quantizer 907, 909 ... Inverse normalization circuit 904 ... Adder

Claims (6)

[Claims] 1. A digital audio signal transmitting apparatus for compressing and encoding digital audio signals of a plurality of channels (n channels, n is a positive integer greater than 3) and transmitting the digital audio signals of the n channels. Of the digital audio signals of m (n> m) channels, which have a higher audible influence than the digital audio signals of the other nm channels, the first is applied.
Second compression coding means for compressing and coding by the first coding method with a compression ratio of 1) and a second compression ratio higher than the first compression ratio for the digital audio signals of the n-m channels. Second compression encoding means for performing compression encoding by the second encoding method having a compression rate of, and the compression encoding output from the first compression encoding means and the second compression encoding. A device for transmitting a digital audio signal, characterized in that the compressed and encoded output from the means is transmitted together. 2. The n channels are a center channel, a left channel, a right channel, a surround left channel and a surround right channel, and the n−m channels are a surround left channel and a surround right channel. The digital audio signal transmission device according to claim 1. 3. The n channels are a center channel, a left channel, a right channel, a left center channel, a right center channel, a surround left channel and a surround right channel, and the n−m channels are a surround left channel. 2. The digital audio signal transmission device according to claim 1, wherein the transmission device is a surround right channel. 4. The first encoding method divides an input digital audio signal into a plurality of bands, blocks the digital audio signal of each band into a plurality of samples, and orthogonally transforms spectrum components in each block. ,
This is a method of adaptively compression-coding according to the auditory characteristics, and the second coding method obtains coefficient information obtained by orthogonally transforming an input digital audio signal for each plurality of samples and sub-information related thereto, and obtains each channel. 2. The digital audio signal transmission apparatus according to claim 1, wherein the method is a system for adaptively compressing and coding by allocating the information amount of each channel according to the energy of. 5. The digital audio signal transmission apparatus according to claim 1, wherein the transmission includes recording on a recording medium. 6. A digital audio signal of a plurality of channels (n channels, n is a positive integer greater than 3),
The first code having the first compression ratio for a digital audio signal of m (n> m) channels, which has a higher auditory influence than other digital audio signals of nm channels. The second encoding method has a second compression rate higher than the first compression rate with respect to the digital audio signals of the n−m channels while being compression-encoded by the encoding method. A receiving device for a digital audio signal, which receives a signal encoded by compression, wherein the first encoding is performed on the digital audio signals of the m channels among the digital audio signals of the n channels. A first decoding means for performing a first decompression decoding corresponding to the system, and a second decoding system for the digital audio signals of the nm channels. That the receiving apparatus in a digital audio signal and having a second decoding means for performing a second decompression decoding.