JPH07121194A - High efficiency encoding device and interface device - Google Patents

Abstract

PURPOSE:To provide a multi-channel digital audio signal with high tone quality without changing an existing format. CONSTITUTION:This device is provided with plural compression-encoding circuits 521-52n compression-encoding respective digital signals of plural channels, a deletion circuit 61 deciding and deleting a redundant channel CH2 incorporating a redundant digital signal and a band division circuit 60 deciding a priority channel CH1 to be made high tone quality when the redundant channel is decided and band dividing the priority channel signal, and when the priority channel CH1 signal is compression-encoded, one side divided by the band division circuit 60 is sent to the compression-encoding circuit 52, corresponding to the priority channel CH1, and the other side is sent to the compression-encoding circuit 52, corresponding to the redundant channel CH2.

Description

Detailed Description of the Invention

[0001]

BACKGROUND OF THE INVENTION 1. Field of the Invention The present invention relates to a high efficiency for compressing and encoding a multi-channel digital signal used in, for example, a stereo such as a motion picture film projection system, a video tape recorder, a video disc player or a so-called multi surround sound system. The present invention relates to an encoding device and an interface device for ensuring compatibility between a multi-channel digital signal and an existing digital interface.

[0002]

2. Description of the Related Art There are various techniques and devices for high-efficiency coding of audio or voice signals. For example, a time domain audio signal is divided into blocks for each unit time, and a signal on the time axis of each block is used. Is converted into a signal on the frequency axis (orthogonal conversion), divided into multiple frequency bands, and coded for each band. A so-called transform coding (transform coating) or time domain An example is band division coding (subband coding: SBC), which is a non-blocking frequency band division method in which an audio signal or the like is not divided into blocks for each unit time and is divided into a plurality of frequency bands for encoding. .
Further, a method and a device for high efficiency coding in which the above band division coding and transform coding are combined are also considered, and in this case, for example, after performing band division by the band division coding, The signal in each band is orthogonally transformed into a signal in the frequency domain, and each orthogonally transformed band is encoded.

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

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

Further, as a frequency division width in the case of quantizing each frequency component divided into frequency bands, for example, there is a band division considering human auditory characteristics. That is, an audio signal may be divided into a plurality of bands (for example, 25 band) with a bandwidth that is generally called a critical band and has a wider bandwidth in a higher band. Further, at the time of encoding the data for each band at this time, encoding is performed by predetermined bit allocation for each band or adaptive bit allocation for each band. For example, when the coefficient data obtained by the MDCT processing is encoded by the bit allocation, M for each block is
The MDCT coefficient data for each band obtained by the DCT process is encoded with an adaptive distribution bit number.

The following two methods and apparatuses are known as the above-mentioned bit allocation method and an apparatus therefor. IEEE Tra
nsactions of Accoustics, Speech, and Signal Processi
In ng, vol.ASSP-25, No.4, August 1977, bit allocation is performed based on the signal size of each band. Also,
ICASSP 1980 The critical band coder--digital encod
ing of the perceptual requirements of the auditory
system MA Kransner MIT describes a method and apparatus that uses auditory masking to obtain a required signal-to-noise ratio for each band and perform fixed bit allocation.

[0007]

Here, in a high-efficiency compression coding system for audio signals using, for example, the above-mentioned sub-band coding, the audio data is approximated by utilizing human auditory characteristics. A method of compressing to 1/5 has already been put into practical use. As a method of compressing this audio data to about 1/5, for example, so-called ATRAC (Adaptive TRansform Acoustic) is used.
There is a method called Coding).

However, in the high-efficiency coding method utilizing the auditory characteristics of human beings, there are some cases in which the musical instrument or human voice obtained by compression coding and then decoding changes from the original sound. In particular, when used in a recording format of a recording medium that requires faithful reproduction of the original sound, it is essential to improve the sound quality.

On the other hand, at present, a high-efficiency coding method (for example, A
The TRAC format) has already been established,
Hardware adopting this format is also spreading. Therefore, a large amount of labor is required to change or expand the format. Therefore, it is desired to achieve high sound quality by modifying the encoding and decoding without changing the format. As a method of improving the sound quality, it is possible to mix linear PCM audio, for example. However, since the frame length and the time length per frame are different between the compressed data of the high-efficiency encoding method and the linear data, it is difficult to synchronize them at the time of reproduction, and the data of these two formats are used at the same time. Is very difficult.

Further, not only in the case of ordinary audio equipment, but also in the case of handling audio or voice signals of a plurality of channels such as a movie film projection system, a video tape recorder, a stereo or multi-surround audio system such as a video disc player. Also,
It is desired to perform high efficiency coding that reduces the bit rate.

Particularly, for example, eight channels of digital audio signals of a left channel, a left center channel, a center channel, a right center channel, a right channel, a surround left channel, a surround right channel and a subwoofer channel are recorded on the above-mentioned motion picture film. In such a case, high-efficiency coding that reduces the bit rate is required. That is, a region capable of recording even the above 8 channels of 16-bit linearly quantized audio data at a sampling frequency of 44.1 kHz as used in so-called CD (compact disc) is secured on the motion picture film. It is difficult to do so, therefore, compression of the audio data is required.

The above 8 recorded on the motion picture film
Each of the channels is, for example, a left speaker, a left center speaker, a center speaker, a right center speaker, a right speaker, a surround left speaker arranged on the screen side on which the image reproduced from the image recording area of the movie film is projected by the projector. It corresponds to a speaker, a surround light speaker, and a subwoofer speaker, respectively. Here, the center speaker is arranged in the center of the screen 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 outputs a reproduced sound based on audio data of the subwoofer channel and effectively outputs a sound felt as vibration rather than a low frequency sound such as an explosion sound. It is often used effectively for such purposes. The left speaker and the right speaker are arranged on the left and right of the screen, and output a 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 is arranged between the left speaker and the center speaker, and the right center speaker is arranged between the center speaker and the right speaker. The left center speaker outputs a reproduced sound based on the audio data of the left center channel, and the right center speaker outputs a reproduced sound based on the audio data of the right center channel. Each of the left center speaker and the right speaker plays an auxiliary role. Fulfill. Especially in a movie theater with a large screen and a large number of seats, the localization of the sound image tends to be unstable depending on the position of the seat, but by adding the above left center speaker and right center speaker, a more realistic localization of the sound image is created. Exert an effect on. Further, the above-mentioned surround left speaker and surround right speaker are arranged so as to surround the spectators' seats, and output the reproduced sound by the audio data of the surround left channel and the reproduced sound by the audio data of the surround right channel. , It has the effect of giving a cheerful impression. As a result, a more stereoscopic sound image can be created.

Further, since the surface of the medium called film is apt to be scratched, if the 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.

From the above, as a compression method for compressing the 8-channel digital audio data, the optimum bit allocation is performed in consideration of the characteristics of human auditory sense as described above.
While compressing 16-bit digital audio data at a sampling frequency of 44.1 kHz such as that recorded on a D (compact disc) to about 1/5, a CD
The high efficiency encoding method (so-called A
The TRAC method, etc.) is applied.

However, in the high-efficiency coding system, since the general musical instrument and the human voice slightly change from the original sound as described above, it is necessary to faithfully reproduce the original sound as in the above movie film. When it is adopted as a recording format of a medium, some means for improving the sound quality is needed. This is a problem that always accompanies the lossy compression from the viewpoint of securing the recording area even if a multi-channel recording format for the motion picture film other than the high efficiency encoding method is used. .

By the way, in recent years, the digitization of audio equipment has progressed, and the digitized equipment has become widespread not only for commercial use but also for consumer use. Especially, for commercial use, multi-channel digital audio is used. This is becoming more common, and for example, equipment that handles the 8-channel digital audio is already in use. Also, the multi-channelization of this digital audio is spreading to consumer use. For this reason, it is necessary to ensure the format compatibility between the various existing digital audio interfaces that handle linearly quantized digital audio samples and the multi-channel audio data. Similarly, it is also necessary to ensure format compatibility between the existing digital audio interface and the multi-channel compression-encoded audio data used in the movie film or the like.

However, the conventional existing digital audio interface, for example, EIAJ (Japan Electronic Machinery Manufacturers Association) CP-1201, aims at transmitting and recording a monaural or stereo audio signal.
There is no scalability for more than one channel.
Therefore, it is necessary to set a new format in order to transmit and record the multi-channel digital audio signal including the compression-coded one as described above. That is, this means that it is not compatible with existing formats. Such incompatibility with the existing format hinders, for example, the development of the above-mentioned multi-channel digital audio into existing consumer devices and recording media. From this fact, it is desired that the existing digital interface and the multi-channel audio including the above compression-coded one are compatible.

Therefore, the present invention provides a high-efficiency coding apparatus capable of improving the sound quality of a multi-channel digital audio signal without changing the existing format.
It is an object of the present invention to provide a format conversion device capable of ensuring compatibility between a current digital audio interface and multi-channel audio simply by adding a simple structure.

[0019]

The high-efficiency coding apparatus of the present invention has been proposed to achieve the above-mentioned object, and uses a predetermined number of allocated bits for each digital signal of a plurality of channels. A plurality of compression-encoding means for compression-encoding, and a redundant channel determination means for determining at least one channel including the redundant digital signal as a redundant channel when there is a channel including the redundant digital signal among the plurality of channels. , A priority channel determining means for determining at least one channel excluding the redundant channel from the plurality of channels as a priority channel when the redundant channel is determined, the compression coding of the digital signal of the priority channel is performed. On the predetermined number of allocated bits corresponding to the priority channel and on the redundant channel It is obtained to perform using a predetermined number of allocated bits.

Here, the compression coding means divides the digital signal of each channel into a plurality of frequency bands, obtains the characteristic of the digital signal for each channel, and determines the characteristic of each channel according to the characteristic of the digital signal. The compression coding is performed by distributing the allocated number of bits for each band. The priority channel determining means comprises a dividing means for dividing the digital signal of the priority channel into a plurality of frequency bands,
The output of the dividing means is sent to the compression coding means corresponding to the priority channel and the compression coding means corresponding to the redundant channel. The redundant channel determining means is provided with deleting means for deleting the digital signal of the redundant channel,
The deleting means deletes the digital signal of the redundant channel before compression coding.

Next, the high-efficiency decoding apparatus corresponding to the high-efficiency encoding apparatus of the present invention expands and decodes the respective digital signals of the plurality of channels which are respectively subjected to the predetermined compression coding processing. Decompression / decoding means and, when there is a channel including a digital signal of the same origin as the digital signal of the priority channel among the plurality of channels, a synthesizing means for synthesizing the digital signal of the same origin into the digital signal of the priority channel. I have.

Here, the synthesizing means frequency-synthesizes the digital signal of the priority channel and the digital signal of the same origin. Further, when the digital signal of the same origin is combined with the digital signal of the priority channel, a digital signal generating means for generating a predetermined digital signal as a digital signal of the channel having the digital signal of the same origin is provided.

Further, the interface device of the present invention is
Digital data of a plurality of channels in which a predetermined compression coding process is applied to a data area of a format having at least a data area of digital data subjected to predetermined signal processing (for example, linear quantization processing) and an information area other than the data area. A predetermined signal inserted and / or inserted into the data area of a signal having a format having at least a data area of digital data that has been subjected to predetermined signal processing (for example, linear quantization processing) and an information area other than that. The format conversion means for separating the digital signals of the plurality of channels, which have been subjected to the compression encoding processing, from each channel.

[0024]

According to the high efficiency coding apparatus of the present invention, when there is a redundant channel containing a redundant digital signal among a plurality of channels, the digital signal of the redundant channel is deleted and compression coding of the digital signal of the redundant channel is performed. The number of allocated bits, which should have been used for the above, is used for compression coding of the digital signal of the priority channel for which high quality compression coding is desired.

Further, according to the interface device of the present invention, predetermined compression encoding is performed on the data area of the existing format having at least the data area of the digital data subjected to the predetermined signal processing and the information area other than the digital data. The processed digital signals of the plurality of channels are inserted and / or the digital signals of the plurality of channels, which are inserted into the data area of the existing format and are subjected to the predetermined compression encoding processing, are separated for each channel. This ensures compatibility with existing formats.

[0026]

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

The high efficiency coding apparatus according to the embodiment of the present invention is shown in FIG.
As shown in FIG. 5, a plurality of compression coding circuits 52 _{1 to} 52 _n for compressing and coding each digital signal of the plurality of channels CH1 to CHn using a predetermined allocation bit number, and redundant channels of the plurality of channels CH1 to CHn. Redundant channel determining means for determining at least one channel including the redundant digital signal as a redundant channel when there is a channel including the digital signal (channel CH2 in the example of FIG. 1), and a plurality of the plurality of channels when the redundant channel is determined. Channel CH1 to CHn of at least one channel (channel CH1 in the example of FIG. 1) excluding the redundant channel CH2 as a priority channel, and a compression code of the digital signal of the priority channel CH1. The above-mentioned predetermined allocation corresponding to the priority channel CH1 Wattage and the redundant channel CH2
Is performed by using the predetermined number of allocated bits corresponding to.

Here, the compression coding circuits 52 _{1 to} 52
_As will be described later, _n is obtained by dividing the digital signal of each channel CH1 to CHn into a plurality of frequency bands, obtaining the characteristic of the digital signal for each channel CH1 to CHn, and determining each channel according to the characteristic of the digital signal. CH
The above-mentioned compression coding is performed by distributing the above-mentioned allocated bits of 1 to CHn for each band. Further, as shown in FIG. 1, the priority channel determining means divides the digital signal of the priority channel CH1 into a plurality of frequency bands by a band division circuit 6
0, the one output divided by the band division circuit 60 is sent to the compression encoding circuit 52 ₁ corresponding to the priority channel CH1, and the other output is transmitted to the compression encoding circuit 52 ₂ corresponding to the redundant channel CH2. Send to. Further, the redundant channel determining means is provided with a deletion circuit 61 for deleting the digital signal of the redundant channel CH2, and the deletion circuit 61 deletes the digital signal of the redundant channel CH2 before sending it to the compression coding circuit 52 _2. To do.

Next, the high-efficiency decoding apparatus corresponding to the high-efficiency encoding apparatus of the above-described embodiment of the present invention respectively performs predetermined compression encoding processing (compression encoding by each of the compression encoding circuits 52 _{1 to} 52 _n ). A plurality of channels CH1 to
A plurality of decompression / decoding circuits 54 _{1 to} 54 _n for decompressing / decoding each digital signal of CHn, and a channel including a digital signal of the same origin as the digital signal of the priority channel CH1 among the plurality of channels CH1 to CHn (see FIG. In the first example, when there is a decoding device side channel CH2) in which the digital signal of the priority channel CH1 is separated and compression-encoded on the encoding device side, the same origin digital signal is converted into the digital signal of the priority channel CH1. And a synthesizing circuit 62 for synthesizing.

Here, the synthesizing circuit 62 frequency-synthesizes the digital signal of the priority channel CH1 and the digital signal of the same origin. Further, when the digital signal of the same origin is combined with the digital signal of the priority channel CH1, a predetermined digital signal (silence in this embodiment is used as the digital signal of the channel (CH2) having the digital signal of the same origin. Digital signal generating means for generating data (silent data generating circuit 6)
3) is provided.

First, before explaining the specific structure of the high-efficiency coding apparatus of the embodiment of the present invention, the inter-channel compression and frequency division recording method adopted in the apparatus of this embodiment will be explained. The voice signal in the embodiment of the present invention includes not only the voice but also various sounds and music signals.

The audio compression technique is a method of converting an input audio signal into frequency components or frequency-dividing and compressing. For example, in the case of a voice compression coding method (subband coding) using the subband division method, the dynamic range is reduced by dividing the frequency band of the input voice signal to reduce uneven distribution of signal energy in each band. However, compressed data is obtained by allocating bits corresponding to the signal energy of each subband and performing encoding.

Therefore, for example, when compression is performed by limiting the frequency band and increasing the number of bits allocated only to the specific frequency component due to the limitation, the compressed data of the specific frequency component is obtained by the same compression method. In addition, the sound quality is improved compared to the compressed data in all frequency bands. This is the basic principle of the encoding of the present invention.

Further, in this embodiment, audio signals of a plurality of channels are handled. For example, for a movie film, a plurality of channels of audio signals are recorded (for example, 8
Channel audio recording) is assumed. Here, in a medium called a movie, there are cases where there is a channel that has no sound in the production, and there is a case where there is sound only in a specific channel. Therefore, the audio data area on the film of the channel in which the audio signal does not exist in this case is redundant, and no problem occurs even if the area is used for improving the sound quality of other channels. That is,
In the above-mentioned multi-channel audio, there may be a channel in which no audio signal exists. Instead of recording the audio data of this redundant channel on the film, use the relevant area for recording for the specific priority channel. By doing so, it is possible to realize high-quality sound recording of the specific channel with priority.

A specific configuration for realizing the above-described inter-channel compression and frequency division recording method is the configuration shown in FIG. Hereinafter, FIG. 1 will be described.

First, the recording system will be described. That is, in FIG. 1, the input terminal 51 ₁ to 51 _n, n-number of channels of the first through n corresponding respectively CH1~CH
n digital audio signals are provided. Also, this figure 1
The compression coding circuit (encoder) 52 ₁ to 52 _n of the provided corresponding to the respective channels CH1 through CHn, these in each compression encoding circuit 52 ₁ to 52 _n, using the sub-band coating, for example to be described later Method of compressing and encoding a signal to about 1/5 (for example, so-called ATRAC method)
The compression encoding of the supplied digital signal is carried out by.

Here, for example, it is assumed that the digital audio signal of the second channel CH2 to the input terminal 51 ₂ is a digital signal of a redundant channel such that the audio signal does not exist, and the digital audio signal to the input terminal 51 ₁ is Channel C of 1
It is assumed that the digital audio signal of H1 is the digital signal of the priority channel for which the high sound quality is desired.

The first channel C which is the above-mentioned priority channel
The H1 digital signal is divided into a plurality of frequency bands by the band division circuit 60 before compression encoding.
Note that the algorithm for the band division, takes the best method depending on the compression method used in the compression encoding circuit 52 ₁ to 52 _n.

On the other hand, the digital audio signal of the redundant channel (second channel CH2) is deleted by the deleting circuit 61 and is not sent to the compression coding circuit 52 ₂ originally provided for the redundant channel CH2. Is done like this.

At this time, the band division circuit 60
Separate the signals of each band of the priority channel and
Originally set for the first channel CH1 which is the destination channel
Compression coding circuit 52₁And send the other to the above
The compression encoding times provided for the second channel CH2
Road 52₂Send to. Therefore, the priority channel
The digital audio signal of the₁
And compression encoding circuit 52 ₂Depending on the number of allocated bits of both
The compression coding is performed (that is, the data of the priority channel is
The number of allocated bits for the digital signal is
Therefore, the other
Higher sound quality can be achieved compared to the channel.

As a method of separating each band of the signal of the priority channel in the band division circuit 60, for example, it is separated into a high band and a low band, a low band and a mid-high band, or a low mid band and a high band. It is possible to consider separation into regions, or separation according to human auditory characteristics. Here, when performing separation according to human auditory characteristics, it is preferable to divide into a band most sensitive to human hearing and a band other than that. By performing such separation according to the human auditory characteristics, it becomes possible to allocate more allocated bits in the subsequent compression coding to the most sensitive band in the auditory sense, and adapted to the human auditory characteristics. , Higher sound quality can be achieved.

The digital audio signals of the other channels are sent to the compression encoding circuits provided correspondingly, and are respectively compression encoded.

The respective compressed data from the respective compression encoding circuits 52 _{1 to} 52 _n are recorded on the recording medium 53 (for example, audio data recording area of movie film). At this time, the compressed audio data of the priority channel is the first
Are recorded in the recording area for the first channel and the recording area for the second channel. The compressed audio data of the other channels are recorded in the corresponding recording areas.

Next, in the reproducing system, the recording medium 53 is used.
The compressed audio data extracted from the audio data recording area for each channel of the (movie film) is respectively corresponded to by the expansion / decoding circuits (decoders) 54 _{1 to} 54 _n provided corresponding to the respective channels. Decompression / decoding similar to the conventional one corresponding to the compression / encoding in the compression / encoding circuits 52 _{1 to} 52 _n is performed.

Here, the decompression-decoded digital audio signals of the channels other than the priority channel are output from the corresponding output terminals 55, and the first and second channels CH1 are used as the priority channels. , CH2
Are recorded in the recording area for recording, and the expansion decoding circuit 54 respectively
_The voice signal decompressed and decoded in ₁ and 54 ₂ is sent to the synthesis circuit 62. The synthesis circuit 62 performs band synthesis corresponding to the band separation in the separation circuit 60. The output of the synthesis circuit 62 is the priority channel, that is, the first channel.
Is taken out from the output terminal 55 ₁ as a digital audio signal of the channel CH1.

On the other hand, for the second channel CH2,
Since no audio signal was present at the time of input to the encoding device, the silence data creating circuit 6 is provided on the decoding device side.
3 generates a silent signal and outputs the silent signal to the second
The signal is output from the output terminal 55 ₂ as a signal of the channel CH2.

Note that FIG. 1 illustrates the case where the first channel CH1 is the priority channel and the second channel CH is the redundant channel. Therefore, the separation circuit 60 and the combining circuit 62 are provided on the first channel side. Although an example is shown in which the deletion circuit 61 and the silence data creation circuit 63 are provided on the side of the second channel, the device of the present embodiment is, for example, the separation circuit, the combination circuit, the deletion circuit, and the deletion circuit for each channel. It is also possible to provide a silence data creation circuit, determine whether or not an audio signal exists in each of the channels, for example, the separation circuit and the deletion circuit, and switch the operation depending on the presence or absence of the audio signal. is there. In this case, the redundant channel and the priority channel are paired, and conversely, when there is no redundant channel, the priority channel is not set. Further, the priority channels are set in the order of the channels in which high quality sound is desired.

Further, it is possible to set one priority channel for a plurality of redundant channels. For example,
To separate the data of one priority channel into a plurality of bands and to use these compression coding circuits for a plurality of redundant channels (including the compression coding circuit for the priority channel) to achieve higher sound quality. You can

In this case, when only the compression side portion corresponding to the priority channel in the configuration of FIG. 1 is extracted and shown, FIG.
Can be expressed as In addition, also in FIG. 2, the first
The case where the above channel is set as the priority channel is taken as an example.

That is, in FIG. 2, when one priority channel is set for a plurality of redundant channels, for example, the priority channel data (PCM data) from the input terminal 51 _{1 of} FIG. Division processing circuit 7
0 (basically the same as the band division circuit 60)
Sent to. The frequency band division processing circuit 70 is composed of, for example, a band division filter, divides the data of the priority channel into a plurality of frequency bands (m frequency bands in the example of FIG. 2), and PCMs of these frequency bands. The data is converted into m corresponding compression coding circuits 52 ₁
Send to ~ 52 _m . These compression encoding circuits 52 _{1 to} 52 _m
Compressed data in compressed is recorded on the recording medium is output through the terminals 73 ₁ to 73 _m (or transmitted to the transmission system)
To be done. In the example of FIG. 2, the description of the redundant channels is omitted, but the number of band divisions in the frequency band division processing circuit 70 is m−1 (from the division number m to one of the original priority channels). Redundant channels) are set, and data is deleted by the deletion circuit in these redundant channels.

The structure on the decompression side corresponding to the example of FIG.
As shown in FIG. In FIG. 3, the compressed data supplied from the recording medium (or via the transmission system) corresponds to the terminals 81 _{1 to} 81 _m corresponding to the m channels.
Through the corresponding decompression / decoding circuits 54 _{1 to} 54.
Decompressed by _m , the decompressed PCM data is sent to the signal combining processing circuit 90 (basically the same as the combining circuit 62). The signal synthesizing processing circuit 90 is composed of a band synthesizing filter and synthesizes the PCM data of each divided band of the priority channel. The PCM data output from the signal synthesis processing circuit 90 becomes band-combined priority channel data and is output from the terminal 55 ₁ . In the example of FIG. 3 as well, the description of the redundant channels is omitted, but m-1 of the number of band combinations in the signal combination processing circuit 90 (one of the original priority channels is divided from the number of bands m). In the redundant channels (excluding the number), the silent data generation circuit generates silent data.

In the apparatus of this embodiment, it is also possible to set one redundant channel for a plurality of priority channels. In this case, for example, each data separated by the plurality of priority channels is The data is sent to the compression coding circuit corresponding to the one redundant channel, and the allocated bits in the compression coding circuit are further divided for each data separated from the plurality of priority channels. In addition, the channel to be the redundant channel and the channel to be the priority channel can be set in advance. In this case, the channel that will be the redundant channel is considered to be of particularly low importance, and the channel is set as the redundant channel only when this channel becomes silent. Try to set the high channel.

On the other hand, when there is no redundant channel such that no sound is produced (voice data does not exist), normal multi-channel audio recording is performed as shown in FIG. In FIG. 4, the same components as those in FIG. 1 are designated by the same reference numerals.

In FIG. 4, only the portion corresponding to the signal flow in the case of normal multi-channel audio recording is shown. In FIG. 4, the input terminals 51 _{1 to} 51 ₁
digital signals of the channels CH1~CHn supplied to _n each corresponding compression encoding circuit 52 ₁ to 52
It is compression-encoded with _n and further recorded in the corresponding recording areas of the recording medium 53. After that, the signals read from the respective recording areas are expanded and decoded by the decompression / decoding circuits 54 _{1 to} 54 _n.
The data is decompressed and decoded by and output from the output terminals 55 _{1 to} 55 _n .

In FIG. 4, only the portion corresponding to the signal flow in the case of normal multi-channel audio recording is shown. Explaining in correspondence with FIG. 1, in the configuration of FIG. When audio recording is performed, the division circuit and the deletion circuit do not perform the division processing and the deletion processing, and the supplied channel data is passed through and sent to the corresponding compression encoding circuits, and the decompression decoding is performed. The output data of the circuit is also sent to the corresponding output terminal as it is.

From the above, the compression method applied to the apparatus of the present invention has the following features. First
In addition, both subband coding and transform coding are effective. Second, there is no need to change the encoder and decoder.

Third, coexistence with conventional compressed data is possible. Fourth, the operation of preferentially allocating bits to a specific band is easy. Fifth, it is effective for multi-channel audio of two or more channels.

Further, each of the first to fifth characteristics will be described in detail. First, in the first feature, the compression method using frequency band division, such as subband coding and transform coding, is effective in that the compression method itself in the compression encoding circuit is a filter (QMF (Quadrature Mirror Filter). ) And PF
This is because the B (Polyphase Filter Bank)) is used to divide the input signal into a plurality of frequency bands and the information is compressed. Therefore, even if the division (the division by the band division circuit) is performed before the compression encoding as in the present embodiment, the harmful effect due to the division does not occur and the sound quality can be improved.

In the second feature, the encoder is optimal for frequency division using a division filter (subband division filter) used internally during encoding.
Immediately before the band division filter (that is, the band division circuit)
It is possible to realize exactly the same by adding. The operations of the pre-encoder (the processing in the band division circuit and the processing in the deletion circuit) are only the determination of the redundant channel in the deletion circuit, the determination of the priority channel in the division circuit, and the frequency division work of the priority channel in the band division circuit. In addition, the decoder is optimal to synthesize using a synthesis filter (subband synthesis filter) used internally during decoding, but immediately after that, a filter (that is, the synthesis circuit) that synthesizes a plurality of audio data is added. Also, each data output can be restored in the same way.
The work of this post-decoder (the processing in the synthesizing circuit and the processing in the silence data creating circuit) is only to synthesize the priority channels in the synthesizing circuit and to create the silence data for the redundant channels in the silence data creating circuit. In both of these pre-encoders and post-decoders, there are few additional components, and the division filter in the band division circuit can be the same as the sub-band division filter in the encoder. The same one as the internal subband synthesis filter can be used.

In the third feature, the frequency band of the data sent to the compression coding circuit is only converted by the band division circuit, and the format of the data compressed by the compression coding circuit is the same as the conventional one. Since it does not change at all, it is possible to record according to the existing recording format.

In the fourth feature, when encoding, it is possible to preferentially allocate bits to a frequency band in which deterioration of sound quality is desired to be prevented. Further, each frequency band to be divided and the number of divisions can be selected at the time of encoding, and the decoder is not affected by the selected contents.

In the fifth feature, in the case of this embodiment,
Since data for one channel (priority channel) must occupy a data area for two or more channels, monaural audio cannot be realized, but in multi-channel audio recording of two or more channels, areas for channels other than the priority channel If the sound can be used due to silence or unused, high sound quality can be achieved.

Further, according to the apparatus of the embodiment of the present invention, the operation mode as shown in FIG. 1 is called a high sound quality mode, and the operation mode as shown in FIG. 2 is called a normal mode. The following applications are possible. First,
For example, in 2-channel audio recording, a stereo normal mode and a monaural high-quality sound mode can be provided.

Secondly, for example, in 4-channel audio recording, it is possible to equip the normal mode for the so-called 3-1 system and the high-quality stereo mode. Thirdly, a high sound quality mode can be provided in a 2-channel n-track (n is a natural number) multi-channel audio recording. In addition, it is also possible to give priority to a specific channel and use a region of three or more channels to further improve the sound quality, or to apply the high sound quality mode to a plurality of channels when the number of media channels is large. is there.

From the above, the compression method of the embodiment of the present invention is a method for improving the sound quality, which is particularly effective when a multi-channel audio signal is recorded on a motion picture film. In the 8-channel audio recording used for the above-mentioned motion picture film, among the eight channels of the left channel, left center channel, center channel, right center channel, right channel, surround left channel, surround right channel and subwoofer channel, The subwoofer channel, which is a channel intended to output audio data of only the frequency component, has a short data section that is characteristically sounded, so there are many sections that can be judged as the redundant channel, and
For example, the center channel can be selected as the priority channel for which high sound quality is desired.

Further, according to the compression method in the embodiment of the present invention, it is easy to support the existing format, and the information indicating whether the data recorded on the recording medium (cinema film) is in the high sound quality mode or not. Is not necessary. That is, in the high sound quality mode, since the data of a specific channel (priority channel) is spread over a plurality of recording areas, information of which channel the data recorded in each recording area records Because it is more appropriate to save. Note that the amount of information that increases in each recording unit by storing information on which channel data is recorded in each recording area is n * log ₂ (n) [bits], where n is the total number of channels. However, the increase in data when recording is negligible.

By the way, in the above-described apparatus of the present invention, in order to make the multi-channel compressed audio data compatible with various existing digital audio interface formats for handling linearly quantized digital audio samples, I am trying to convert the format like.

That is, in the format conversion of the present embodiment, in order to ensure compatibility with the existing digital audio interface, the inside of the interface format is not entered. For this reason, a new frame is provided only in the data section of the existing format so that the data can be stored again. In other words, in the format conversion of this embodiment, only the areas allocated for recording and transmission of linearly quantized audio samples in various existing digital audio formats are changed. At this time, for example, it can be distinguished from normal audio data by using the broadcasting station studio channel status c1 = 1 (mode other than audio) defined by the CP-1201. At this time, in this embodiment, in order to secure the area for multi-channel,
The handling within the format for recording and transmission shall be stereo audio signals.

As described above, in this embodiment, since the multi-channel audio data is stored in the area of the stereo audio signal, the audio data is compressed. For example, the audio data can be compressed to about 1 / 4.8 according to the method of efficiently compressing and coding the audio signal by using the human auditory characteristic by using the subband coding or the like. Up to 9 channels of audio data can be stored in the stereo area of the existing format. The value recorded in the sampling frequency in the recording and transmitting format is a value according to the compression method used. For example, when the high efficiency coding method is used, the frequency is always 44.1 kHz. Also, if the maximum number of quantization bits is 24 bits, the area that can be secured will increase accordingly, but so-called CD (compact disc) and DA
Considering the compatibility with T (digital audio tape) and the compression method, it is 16 bits. This can be extended.

Hereinafter, description will be made with reference to FIGS. In the conversion format of the present embodiment, the size of one frame is 512, which is the larger because the processing unit of linearly quantized data in a typical compression method of digital audio is 384 samples, 256 samples, or 512 samples. Select a sample. When using a compression method of 384 samples or the like, it is necessary to devise a separate format.

Therefore, one frame has 512 samples of 2048 bytes, and as shown in FIG. 5, a data section 204 for recording data and a code section (202 to 203) for recording information on the data are stored in this area. Can be set. For example, in the code section, a sync code area 201 for synchronizing frames, a control code area 202 for recording data management information, and a sub information area 20 for recording other information.
3 areas of 3 are secured. If this is recorded in the right-justified order to change the amount of information, it is possible to freely record various data. Similarly, since the data section 204 has a variable data size, the data section 204 may be recorded in a rear-justified manner.

Here, the sync code area 201 is
Required for use when synchronizing frames.
The area size of the sync code area 201 may be 4 bytes, and the sync code is represented by hexadecimal 7379, for example.
6e63 or the like is preferable. Further, since the format does not limit the internal format of the data, the same data as this sync code may appear outside the sync.
Therefore, for the sync discrimination, it is necessary to use, for example, one that appears every 2048 bytes as a sync.

The control code area 202 is an area for recording a data control code. The sub information area 203 also records the control code, but the control code area 202 is always present and is different in size from the sub information area 203. The control code area 202 can be divided into, for example, five areas as shown in FIG.

In FIG. 6, the channel count area 210 is an area for recording the number of data channels.
The size of the channel count area 210 is 1 byte, and in order to cope with an increase in the number of channels in the future, it is possible to arbitrarily set 0x00 to 0xff in the format.

The data information area 211 is
This is an area for recording information of recording data. As shown in FIG. 7, the data information area 211 has at least three areas 221 for recording the coding method.
Bits and 2 bits of the area 222 for recording the sampling frequency are reserved. In addition to these areas 221, 222, a reserve area 220 of 3 bits is prepared in the data information area 211.

The data size area 212 is an area for recording the area size of the data area. The area size of the data size area 212 is 2 bytes, and it is possible to record from 0x0000 to 0xffff in the format, but in reality, due to the frame size restriction, 0x0000 to 0x0.
It does not fall outside the range of 7f4.

Sub-information size area 213
Is an area for recording the area size of the sub-information area 203. The area size of the sub-information size area 213 is 2 bytes and is 0x as described above.
It does not fall outside the range of 0000 to 0x07f4.
The size of the sub-information size area 213 and the data size area 212 is 0x.
It does not fall outside the range of 0000 to 0x07f4.

Sub information mode area 214
Is an area for recording a flag designating a sub information mode. The area size of the sub information mode area 214 is 2 bytes. If the mode is specified here, the sub information area 2
In 03, management information can be freely recorded.

The data area 204 of FIG. 5 is an area for recording multi-channel digital audio data.
There is no problem in the format even if the content is not voice. Data is recorded in the last position, and sub information area 2
Between 0 and 3 is filled with 0 data.

The above format is an example, but by setting such a frame format, it is possible to handle multi-channel data in an integrated manner.

[0081] That is, in the specific configuration of the format conversion, as shown in FIG. 8, the input terminals 31 ₁ to
The multi-channel audio compressed data including the first to nth channels supplied via 31 _n are converted into the formats shown in FIGS. 5 to 7 by the format conversion post encoder 30 and output from the output terminal 32. To do. Thereby, compatibility between the multi-channel audio compressed data and the existing digital audio interface can be ensured.

Further, as shown in FIG. 9, the data into which the multi-channel audio data compressed into the existing digital audio format from the terminal 36 is inserted is converted by the format conversion predecoder 35 into the first to first data.
The multi-channel compressed data of the nth channel is taken out and output via the corresponding output terminals 37 _{1 to} 37 _n . Thereby, compatibility between the multi-channel audio compressed data and the existing digital audio interface can be achieved.

Existing digital audio recording media and transmission systems can be made compatible with multi-channel digital audio simply by adding the two format converting means shown in FIGS. 8 and 9.

There is an 8-channel DAT (digital audio tape) as an example of converting a signal from an existing medium into a multi-channel by using the format conversion of the present embodiment. This is a technique of recording 8-channel compressed audio data in DAT by using the format conversion of the present embodiment, and by using this technique, 8-channel compressed audio data from a movie film can also be recorded in DAT. The recording medium is not limited to the above-mentioned DAT, and there is a sampling frequency mode of 44.1 kHz such as a CD (compact disc) and a video disc.
Any 6-bit linearly quantized audio sample can be used as long as it is a digital recording medium in which recording and transmission areas are secured.

FIG. 10 shows an example of multi-channel DAT synchronized with video.

In FIG. 10, the format conversion predecoder 35 starts the digital audio tape recorder 42 when receiving the CUE signal from the video device 40 such as a projector or a projector. At this time,
A sync signal and external control are sent from the predecoder 35 to the digital audio tape recorder 42,
From the digital audio tape recorder 42,
Data in ES / EBU format is provided. The predecoder 35 decodes the AES / EBU format data into multi-channel data and sends the multi-channel compressed data to the decoder 41 for decoding the multi-channel audio compressed data as described above. Note that the digital tape recorder 42 can find the beginning of the recorded time code,
Therefore, it can be applied as a backup system of an apparatus that handles the above-mentioned 8-channel compressed audio data.

Further, taking advantage of the fact that the present invention can be applied to the existing digital recording medium, the compressed audio 8 channel data is recorded on the video disk, and the number of channels (or the number of connected speakers) which can be reproduced by the reproducing device side is used. ), The development such as the development of a device that selects and outputs a sound is considered.

Further, in the transmission according to the present invention,
With the existing digital transmission cable, if the compression method of about 1/5 (for example, the so-called ATRAC method) is used, transmission of up to 9 channels becomes possible. Further, it is possible to discriminate the existing 2-channel mode depending on the format of the transmission interface, and the interface can be shared with the existing digital audio data.

The format conversion of this embodiment is a format in which the size of the sub-information area and the data area can be changed, and the compression method is not limited. Therefore, in the future, by using a new coding method, it is possible to support more channels,
It is also possible to record and transmit a voice that achieves higher sound quality. Further, according to the present embodiment, the data size can be changed for each frame, and it is possible to correspond to the variable-rate or variable-channel compression encoding method.

As described above, the high-efficiency coding apparatus according to the present invention and the high-efficiency decoding apparatus according to the present embodiment require the encoder and the decoder to be changed in the audio compression using the existing subband coding. It is possible to improve the sound quality of a specific channel (priority channel) without using the recording areas of other channels. That is, in the encoder, the input voice is first frequency-divided into a plurality of frequencies and then each is encoded. As a result, since the frequency band of voice is limited for each data, the number of recording bits allocated to each frequency is relatively increased and high sound quality is achieved. At this time,
Since subband coding itself uses frequency division, there is little possibility of problems due to division, and if the division method is devised, it is easy to allocate many bits to the frequency band that needs to prevent deterioration. is there. Also,
In the decoder, each of the divided encoded data may be decoded based on each compression method, and the output data may be combined with each other. Therefore, not only the decoder need not be changed, but also the conventional data and the high sound quality data can coexist.

Further, according to the interface device of the embodiment of the present invention, the multi-channel compressed audio data can be transmitted, recorded and reproduced by using the transmission and recording format of the existing 2-channel digital audio interface. This allows, for example, existing digital 2
It is possible to transmit, record and reproduce multi-channel compressed audio data by utilizing channel transmission and recording formats. For example, by using a compression encoding method that compresses data to about 1/5 using subband coding, It is possible to support up to 9 channels in the digital 2 channel transmission and recording format.

Next, the compression coding circuit 5 shown in FIG.
A specific configuration of 2 _{1 to} 52 _n will be described. In these compression encoding circuits 52 _{1 to} 52 _n , audio PCM is used.
An input digital signal such as a signal is subjected to band division encoding (SB
C), adaptive transform coding (ATC), and adaptive bit allocation (APC-AB).

Hereinafter, description will be made with reference to FIG. In the compression encoding circuit of the present embodiment shown in FIG. 11, 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 of the frequency axis. , The so-called critical bandwidth (critical band) in consideration of human auditory characteristics described later
Each bit is adaptively allocated and encoded. This time,
In the high range, a band obtained by further dividing the critical bandwidth is used. 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. This critical band is a frequency band divided in consideration of human auditory characteristics, and is the band of a pure tone when the pure tone is masked by narrow band noise of the same strength near the frequency of that pure tone. That is. 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. 11, the input terminal 10
Is supplied with an audio PCM signal of 0 to 22 kHz, for example. This input signal is, for example, a so-called QMF.
Is divided into a band of 0 to 11 kHz and a band of 11 to 22 kHz by a band dividing filter 11 such as
Similarly, the band signal is 0 to 5.5 kHz band and 5.5.k to 11 kHz by the band dividing filter 12 such as so-called QMF.
It is divided into a band and. 11 from the band division filter 11
The signal in the k to 22 kHz band is sent to an MDCT (Modified Discrete Cosine Transform) circuit 13 which is an example of an orthogonal transform circuit, and 5.5 k to from the band division filter 12 is transmitted.
The signal in the 11 kHz band is sent to the MDCT circuit 14, and the signal in the 0 to 5.5 kHz band from the band division filter 12 is sent to the MDCT circuit 15.
CT processed. In addition, each MDCT circuit 13, 14, 1
5, block determination circuits 19 and 2 provided for each band
M based on the block size determined by 0, 21
DCT processing is performed.

Here, the block decision circuits 19 and 2 are
MDCT circuits 13 and 14 determined by 0 and 21
A specific example of the block size in 15 is shown in A and B of FIG. 12A shows the case where the orthogonal transform block size is long (the orthogonal transform block size in the long mode), and FIG. 12B shows the case where the orthogonal transform block size is short (the orthogonal transform block size in the short mode). is doing. In the specific example of FIG. 12,
Each of the three filter outputs 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 middle band of 5.5 to 11 kHz, if the block length is long (A in FIG. 12), the number of samples in one block is 128 samples. Then, when a short block is selected (B in FIG. 12), the number of samples in one block is a block for every 32 samples. On the other hand, for a signal in the 11 kHz to 22 kHz band on the high frequency side, when the block length is long (A in FIG. 12), the number of samples in one block is 256, and when the short block is selected ( In FIG. 12B), 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. The block decision circuit 1
The information indicating the block size determined in 9, 20, and 21 is the adaptive bit allocation coding circuits 16, 17, and 18 described later.
And output from the output terminals 23, 25 and 27.

Referring again to FIG. 11, each MDCT circuit 1
The spectrum data in the frequency domain or the MDCT coefficient data obtained by MDCT processing at 3, 14, and 15 are combined into so-called critical bands (critical bands) or bands into which the critical band is further divided in the high band, and adaptive bits are collected. It is sent to the distribution coding circuits 16, 17, and 18.

Adaptive bit allocation encoding circuits 16, 17, 1
In 8, the spectrum data (or MDCT coefficient data) according to the information of the block size and the number of bits assigned to each critical band (critical band) or a band obtained by further dividing the critical band in the high band.
Is requantized (normalized and quantized).
The data coded by the adaptive bit allocation coding circuits 16, 17, 18 are taken out via the output terminals 22, 24, 26. Also, the adaptive bit allocation encoding circuit 16,
In 17 and 18, the scale factor indicating what kind of signal magnitude the normalization has been made and the bit length information showing what kind of bit length the quantization has been performed, and these are also output terminals at the same time. It is output from 22, 24 and 26.

Further, each MDCT circuit 1 in FIG.
The outputs of 3, 14, and 15 calculate the energy of each band obtained by further dividing the critical band in the critical band (critical band) or the high band, for example, the square root of the root mean square of each amplitude value in the band. It is required by things. 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.

Here, the adaptive bit allocation circuits 16 and 1
A more specific configuration of Nos. 7 and 18 will be described with reference to FIG. In the adaptive bit allocation circuit shown in FIG. 13, 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 803 for each band
The energy for each band calculated in step S.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 hearing noise level, first obtains the permissible noise amount for each critical band in consideration of the so-called masking effect, etc., based on the spectrum 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 bit thus obtained and the bit dependent on the permissible noise level of the hearing are added, and the adaptive bit allocation encoding circuit 16 of FIG.
17 and 18 requantize each spectrum data (or MDCT coefficient data) according to the number of bits allocated to each critical band or in the high frequency band, which is obtained by further dividing the critical band into a plurality of bands. . The data encoded in this way is taken out through the output terminals 22, 24 and 26 of FIG.

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

FIG. 14 is a block circuit diagram showing a schematic configuration of a specific example in which the above-mentioned allowable noise calculating circuits are collectively described. In FIG. 14, 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. 15 shows such a Bark spectrum SB for each critical band. However, in FIG. 15, in order to simplify the illustration, the number of bands of the critical band is expressed by 12 bands (B1 to B12).

Here, in order to consider the influence of so-called masking of the Bark spectrum SB, a convolution processing 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 subtracts the masking threshold from the Bark spectrum SB, so that the Bark spectrum SB becomes 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 allowable noise correction circuit 530 and the output terminal 531. 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 dependent on the permissible hearing noise level are added, and the adaptive bit allocation coding circuit 1 is used by using the distributed bit number information.
Encoding is performed in 6, 17, and 18.

In summary, in the adaptive bit allocation coding circuits 16, 17, and 18, the energy of each band band (critical band) of the critical band or the energy of a band obtained by further dividing the critical band 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, the data showing the so-called minimum audible curve RC which is the 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.
If it is assumed that the noise in the vicinity is not heard, 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. In addition, FIG. 16 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, at 4 kHz, even if the sound pressure drops by 8 to 10 dB from 1 kHz, it sounds as loud as 1 kHz, and conversely, at around 50 Hz, it must be about 15 dB higher than the sound pressure at 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, the bit amount dividing method will be described.
Returning to FIG. 13, the signal from the input terminal 801 to which the MDCT circuit output is supplied is also given to the spectrum smoothness calculation circuit 808, and the spectrum smoothness is calculated here. 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.

Next, FIG. 17 shows the configuration of the decompression / decoding circuits 4 ₁ to 4 _n shown in FIG.

In FIG. 17, the quantized MDCT coefficients of each band are given to the input terminals 122, 124 and 126, and the used block size information is given to the input terminals 123, 125 and 127. Decoding circuit 11
6, 117 and 118 cancel the bit allocation using the adaptive bit allocation information.

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. After that, band synthesis is performed in the IQMF circuits 111 and 112, and the result is output from the output terminal 110.

[0122]

As described above, in the high efficiency coding apparatus of the present invention, when there is a redundant channel including a redundant digital signal among a plurality of channels, the digital signal of this redundant channel is deleted and the redundant channel Since the number of allocated bits that should have been used for compression encoding of digital signals is used for compression encoding of digital signals of priority channels for which high quality compression encoding is desired, compression algorithm, recording, transmission There is no need to change the format, and by taking advantage of multi-channel, high sound quality can be achieved by adding a simple configuration.

Further, in the interface apparatus of the present invention, the predetermined compression coding processing is performed on the data area of the existing format having at least the data area of the digital data which has been subjected to the predetermined signal processing and the other information area. Insert digital signals of multiple channels,
And / or by separating the digital signals of a plurality of channels, which are inserted into the data area of the existing format and which have been subjected to the predetermined compression encoding processing, for each channel, the existing structure is added only by adding a simple configuration. It is possible to ensure compatibility with the format.

[Brief description of drawings]

FIG. 1 is a diagram for explaining an operation in a high sound quality mode in a high efficiency encoding and decoding device according to an embodiment of the present invention.

FIG. 2 is a block circuit diagram for explaining a configuration in which data of a priority channel is divided into a plurality of bands and compression-encoded.

FIG. 3 is a block circuit diagram for explaining a configuration for decompressing, decoding, and synthesizing data of a priority channel divided into a plurality of bands and compression-coded.

FIG. 4 is a diagram for explaining an operation in a normal mode in the device of the embodiment of the present invention.

FIG. 5 is a diagram for explaining a frame format of a recording format at the time of format conversion of the present embodiment.

FIG. 6 is a diagram for explaining the format of a control code area in the recording format at the time of format conversion of the present embodiment.

FIG. 7 is a diagram for explaining a format of a data information area of a recording format at the time of format conversion of the present embodiment.

FIG. 8 is a diagram showing a format conversion post encoder of the apparatus of the present embodiment.

FIG. 9 is a diagram showing a format conversion predecoder of the device of the present embodiment.

FIG. 10 is a diagram showing an example in which the format conversion of the present embodiment is applied to a video synchronization system.

FIG. 11 is a block circuit diagram showing a configuration example of a compression encoding circuit of the present embodiment.

FIG. 12 is a diagram showing frequency and time division of a signal in the device of this embodiment.

FIG. 13 is a block circuit diagram showing a configuration for implementing a bit allocation method using two parties of the magnitude of an information signal and the permissible noise spectrum of hearing.

FIG. 14 is a block circuit diagram showing a configuration for obtaining an allowable noise level.

FIG. 15 is a diagram showing an example of a masking threshold depending on the signal level of each band.

FIG. 16 is a diagram showing an information spectrum, a masking threshold, and a minimum audible limit.

FIG. 17 is a block circuit diagram showing a configuration example of a decompression decoding circuit of the high efficiency decoding device according to the present embodiment.

[Explanation of symbols]

52 _{1 to} 52 _m , 52 _n ... Compression encoding circuit 53 ... Recording medium 54 _{1 to} 54 _m , 54 _n Decompression decoding circuit 60 ...・ ・ ・ ・ ・ ・ Splitting circuit 61 ・ ・ ・ ・ ・ ・ ・ ・ ・ ・ ・ Delete circuit 62 ・ ・ ・ ・ ・ ・ ・ ・ ・ ・ ・ Combining circuit 63 ・ ・ ・ ・ ・ ・ Silence data Generation circuit 70 ・ ・ ・ ・ ・ ・ ・ ・ ・ ・ Frequency band division processing circuit 90 ・ ・ ・ ・ ・ ・ ・ ・ ・ ・ Signal synthesis processing circuit 30 ・ ・ ・ ・ ・ ・ ・ ・ ・ ・ Format conversion post Encoder 35 ・ ・ ・ ・ ・ ・ ・ ・ ・ ・ Format conversion predecoder 40 ・ ・ ・ ・ ・ ・ ・ ・ ・ ・ Video device 41 ・ ・ ・ ・ ・ ・ ・ ・ ・ ・ Decoder 11, 12 ・ ・ ・... QMF circuit 13, 14, 15 ... MDCT circuit 16, 17, 18 ... Adaptive bit allocation coding circuit 19, 20, 21 ... Block size determination circuit 22, 24, 26 ... Encoding output terminal 23, 25, 27 ... Block size information output terminal 116, 117, 118 ... Adaptive bit allocation decoding circuit 113, 114, 115 ... IMDCT circuit 112, 111 ... IQMF circuit 520 ... Allowable noise calculation circuit 521 ... Allowable noise calculation circuit input terminal 522 ... For each band Energy detection circuit 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 801 ... MDCT circuit output terminal 802 ... Usable total bits Raw 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 807 ... Bit allocation amount of each band Output terminal 808 ... Spectrum smoothness calculation circuit 809 ... Bit division rate determination circuit 811, 812 ... Multiplier

Claims (5)

[Claims] 1. A compression coding means for compressing and coding each digital signal of a plurality of channels using a predetermined number of allocated bits, and when there is a channel including a redundant digital signal among the plurality of channels, the redundancy is provided. Redundant channel determining means for determining at least one channel including a digital signal as a redundant channel, and when the redundant channel is determined, at least one channel excluding the redundant channel from the plurality of channels is determined as a priority channel. Priority channel determining means, and compression coding of the digital signal of the priority channel is performed using the predetermined number of allocated bits corresponding to the priority channel and the predetermined number of allocated bits corresponding to the redundant channel. A high-efficiency coding device characterized by the above. 2. The compression coding means divides the digital signal of each channel into a plurality of frequency bands, obtains the characteristics of the digital signal for each channel, and determines the characteristics of each channel according to the characteristics of the digital signal. 2. The high efficiency coding apparatus according to claim 1, wherein the compression coding is performed by distributing the number of allocated bits for each band. 3. The priority channel determining means comprises a dividing means for dividing the digital signal of the priority channel into a plurality of frequency bands, and the output of the dividing means is sent to the compression encoding means. The high efficiency encoding device according to claim 1. 4. The redundant channel determining means includes deleting means for deleting the digital signal of the redundant channel, and the deleting means deletes the digital signal of the redundant channel before compression encoding. The high efficiency encoding device according to claim 1. 5. A digital signal of a plurality of channels in which a predetermined compression coding process is applied to a data area of a format having at least a data area of digital data subjected to a predetermined signal processing and an information area other than the data area. Inserted and / or subjected to predetermined compression encoding processing inserted into the data area of a signal of a format having at least a data area of digital data subjected to predetermined signal processing and an information area other than that. An interface device comprising format conversion means for separating digital signals of a plurality of channels for each channel.