Encoding method and apparatus, decoding method and apparatus and recording medium.

TITLE OF THE INVENTION

Encoding Method and Apparatus, Decoding Method and Apparatus and Recording Medium BACKGROUND OF THE INVENTION Field of the invention

This invention relates to an encoding method and apparatus for encoding multi-channel acoustic signals, a apparatus for decoding the encoded xording medium having the encoded signals recorded thereon.

Description of Related Art

There has so far been proposed recording medium capable of recording signals such as the encoded acoustic information or the music information (referred to hereinafter as audio signals), such as a magneto-optical disc. Among methods for high-efficiency encoding of the audio signals, there are a socalled transform coding which is a blocking frequency spectrum splitting method of transforming a time-domain signal into frequency domain signals by orthogonal transform and encoding the spectral components from one frequency band to another, and a sub-band encoding (SBC) method, which is a non-blocking frequency spectrum splitting method of splitting the time-domain audio signals into plural frequency bands withojut blocking and encoding the resulting signals of the frequency bands. There is also known a highdecoding method and information and a re

Γ ‘Τ'¹'·' τ τ ι is blocked every pre-s discrete Fourier trans (DCT) or modified DC converting the signals efficiency encoding technique which is a combination of the sub-band coding and transform coding, in which case the time domain signals are split into plural frequency bands by SBC and the resulting band signals are orthogonal transformed into spectral components which are encoded from band to band.

Among the above-mentioned filters is a so-called QMF (Quadrature Mirror Filter) filter as discussed in 1976, R.E. Crochiere, Digital Coding of Speech in subbands, Bell Syst. Tech. J. Vol.55, No.8, 19 76. This QMF filter splits the frequency spectrum into two bands of equal bandwidths and is characterized in that so-called aliasing is not produced on subsequently synthesizing the split bands. The technique of dividing the frequency spectrum is discussed in Joseph H. Rothweiler, Polyphase Quadrature Filters- A New Subband Coding Technique, ICASSP 83 BOSTON. This polyphase quadrature filter is characterized in that the signal can be split at a time into plural bands of equal band-width.

Among the above-mentioned techniques for orthogonal transform is such a technique in which an input audio signal et unit time, such as every frame, and ^?orm (DFT), discrete cosine transform :T (MDCT) is applied to each block for from the time axis to the frequency axis. Discussions of the MDCT are found in J.P. Princen and A.B. Bradley, Subband/Transform coding Using Filter Bank

-Si*’’* <rwi

Based on Time Domain Aliasing Cancellation, ICASSP 1987.

If the above-mentioned DFT or DCT is used as a method for transforming waveform signals into spectral signals, and transform is applied based on a time block composed of M samples, M independent real-number data are obtained. It is noted that, for reduci blocks, a given time ng junction distortions between time bock is usually overlapped with Ml samples with both neighboring blocks, and M real-number data on an average are quantized and encoded in DFT or DCT for > these M real-number data that are and encoded.

d, if the above-mentioned MDCT is orthogonal transform, M independent (M-M1) samples. It i subsequently quantized On the other han used as a method for real-number data are obtained from 2M samples overlapped with M samples of bo MDCT, M real-number th neighboring time blocks. Thus, in data on an average are obtained for M

I samples and subsequently quantized and encoded. A decoding device adds waveform elements obtained on inverse transform in each block from the codes obtained by MDCT with interference for re-constructing the waveform signals.

In general, if a time block for transform is lengthened, the spectrum frequenc^ resolution is improved such that the signal energy is concentrated in specified frequency components. Therefore, by using MDCT in which, by overlapping with one half of each of both neighboring blocks, with long block lengths, and in which lting spectral signals is not increased ' · TT’· transform is carried ou the number of the resu beyond the number of the original time samples, encoding can be carried out with hig is used. Moreover, her efficiency than if the DFT or DCT since the neighboring blocks have a long transform block raise the integration ⁵ sufficiently long overlap with each other, the inter-block distortion of the waveform signals can be reduced. However, if the transform block length for transform is lengthened, more work area is required for transform, thus obstructing reduction in size of reproducing means. In particular, use of at a time point when it is difficult to degree of a semiconductor should be avoided since this increases the manufacturing cost.

By quantizing signals split into plural frequency bands by a filter or orthogonal transform, the frequency band in ¹⁵ which occurs the quantization noise can be controlled so that encoding can be achieved with psychoacoustic higher ing acoustic characteristics such as masking effects. If ihe signal components are normalized t

with the maximum values of the absolute values of the signal ective bands, encoding can be achieved with still higher efficiency.

As frequency band widths in case of quantizing the frequency components, obtained on splitting the frequency spectrum, it is known to split the frequency spectrum in such manner as to take account of the psychoacoustic characteristics of the the audio signals are d luman auditory system. Specifically, ivided into a plurality of, such as 25, bands using bandwidth^ increasing with increasing frequency. These bands are known as critical bands. In encoding the band-based data, encoding is car bit allocation on the band basis.

cessing b ried out by fixed or adaptive

In encoding coefficient data y bit allocation as described obtained by MDCT pro above, encoding is by a n adaptive number of bit allocation for ¹⁰ band-based MDCT coefficients obtained by block-based MDCT processing. As these known the following tw For example, in Transform Coding o bit allocation techniques, there are o techniques.

sky and P. Noll, ‘Adaptive in ‘IEEE

R. Zelin if Speech Signals’ and ¹⁵ Transactions of Acoustics, Speech and Signal Processing, vol. ASSP-25, No.4, August 1977, bL allocation is performed on litude of the band-based signals. With this system, the quantization noise spectrum becomes flat, such that the quantization noise is minimized. However, the actual noise feeling is not psychoacoustically optimum because the psychoacoustic masking effect is not exploited.

In a publication ‘ICASS

1980, ‘The critical band coder— digital encoding of the perceptual requirements of the auditory system, M.A masking mechanism is

Krasner, MIT’, the psycho acoustic ί I used to determine a fixed bit allocation that produces the nec essary signal-to-noise ratio for each critical band. However, if this technique is used to measure characteristics of a sine wave input, non-optimum results are obtained because of the fixed allocation of bits among the critical bands.

For overcoming thbse problems, there is proposed a highefficiency encoding device in which a portion of the total for bit allocation is used for a fixed pre-fixed from one small block to ¹⁰ another and the remaining portion is used for bit allocation dependent on the signal amplitudes of the respective blocks, and in which the bit number division ratio between the fixed bit allocation and the bit allocation dependent on the signal amplitudes is made dependent on a signal related to an input number of bits usable bit allocation pattern signal, such that the bit number division ratio to the fixed bit allocation becomes larger the smoother the signal spectrum.

This technique significantly improves the signal-to-noise ratio on the whole by allocating more bits to a block including a particular signal spectrum exhibiting concentrated signal energy, as in the case of a sine wave input in which signal energies are concentrated in specified spectral signal components. By using the above techniques, for improving the signal-to-noise ratio characteristics, not only the measured values are increased, but also the sound as perceived by the listener is improved in signal quality, because the human sitive to signals having acute spectral auditory system is sen components .

A variety of differ proposed, and a mo mechanism has also perceptually higher e supposing that the ent bit allocation techniques have been del simulating the human auditory become more elaborate, such that neo ding efficiency can be achieved encoding device capability is correspondingly improved.

In these techniques, the customary practice is to find real-number reference values for bit allocation, realizing the signal-to-noise characteristics as found by calculations as faithfully as possible, the reference values as For constructing quantization fineness and to use integer values approximating allocated bit numbers, a real codestring, it suffices if the information and the normalization coefficient information are encoded with pre-set numbers of bits, from one normali2 ation/quantization band to another, and antized spectral signal components are standard (ISO/IFC 1 1 172-3:1993 (E), the normalized and qu encoded. In the ISO

1993), there is described a high-efficiency encoding system in which the numbers of bits representing the quantization fineness information ire set so as to be different from one band to another. Specifically, the number of bits representing the quantization fineness information is set so as to be decreased with the increased frequency.

There is also known a method of determining the quantization fineness information in the decoding device from, i

for example, the normalization coefficient: information. Since ί

the relation between the normalization coefficient information and the quantization fineness information is set at the time of setting the standard, it becomes impossible to introduce the control based on a more advanced in future. In addition, if there is a quantization fineness psychoacoustic model width in the compression ratio necessary to set the relation coefficient information and information from one c ompressio to be realized, it becomes between the normalization the quantization fineness n ratio to another.

There is also known a method of using variable length codes for encoding for realization of more efficient encoding of quantized spectral signal components, as described in D.A.

r Construction of Minimum Redundancy , 40, p. 1 0J98 (1952).

nal Laying-Open WO94/28633 of the

Huffman, A Method fo Codes”, in Proc. I.R.E.

In the Internatio present Assignee, ther perceptually critical osed a method of separating omponents, that is signal energy concentrated in the e is disc:

tonal c components having the signal vicinity of a specified frequency, from the spectral signals, and encoding the signal components separately from the remaining spectral components. This enables audio signals to be efficiently encoded with a high compression ration without ·Τ substantially deteriorating the psychoacoustic sound quality.

ed encoding techniques can be applied of acoustic signals constructed by ample, the encoding techniques can be

The above-describ to respective channels plural channels. For ex applied to each of the left channel associated with a left-side speaker and the right speaker. The L and R c

Transform Coding of Johnston, ICSSSP89, technique of taking a the spectrum of the su masking curve to achie frequently that the Lchannel associated with a right-side liannels of the audio signals are known to be correlated to each other such that this correlation can be used to realize encoding with a higher efficiency. For example, there is described in the publication ‘Perceptual Wideband Stereo Signals’, James D. picture processing. 1 993 to 1995) a sum and a difference of the input L and R channels, converting these into spectral signals ,finding a masking curve from the spectrum of the sum and quantizing im and that of the difference using the :ve encoding. Since in general it occurs •channel signals are similar to the R· channel signals, the signal level of the channel corresponding to (L-R) is rather low such that it an be encoded with a relatively small number of bits. It is an ancillary merit of this technique that monaural signals can be reproduced by reproducing only the signals of the channel corresponding to (L + R).

In the ISO standard (ISO/IEC 1 1 1 72-3 : 1 993 (E), 1993), there is described a svstem in which L and R channels are : ·........

split by a filter, both

L and R channel signals or channel signals corresponding to (L + R) and (L-R) are encoded for the bands of the lower freq coefficients of each separately encoded for uency and in which only normalization band with the higher frequency are both channels .

With the method of encoding the channel corresponding to the (L + R) and to the channel corresponding to (L-R) in the entire spectrum, signals of the two channels need to be encoded even if one of the channels is of low signal level. This sets a limit to improving the compression efficiency by encoding. If only the normalization coefficients are encoded for the high-range side of bits in encoding t entire frequency spectr there is required a significant number ie normalization coefficients for the um.

Moreover, in the above technique, transform and inverse transform operations are required for encoding and decoding for signals of the entire spectrum of each of two channels, thus increasing the volume of the buffer memory used for transient data storage. SUMMARY OF THE IN

It is therefore a provide a encoding me channel acoustic signa decoding the encoded

VENTION n object of the present invention to thod and apparatus for encoding multis, a decoding method and apparatus for information and a recording medium in which the volume of the encoding information can be decreased and in wh|ich encoding and decoding can be achieved with a smaller volume of the buffer memory.

In one aspect, the present invention provides a method for encoding an input generating plural 1 o w range signals smaller i signal derived from plu step of encoding each signal including a splitting step of range signals and a number of high n number than the input signal from a ral input signals, a low-range encoding of the plural low-range side signals, a high-range encoding step of encoding the high-range side signal and for generating the encoding information obtained by the encoding step £.nd a generating step for generating a codestring based on signals encoded by the low-range encoding step and the high-range encoding step, and on the encoding information.

In another aspect, the present invention provides an apparatus for encoding an input signal including splitting means for generating plural low-range signals and a number of high-range signals smaller in number than the input signal from plural input signals, low-range ncoding each of the plural low-range ge encoding means for encoding the il and for generating the encoding from a signal derived encoding means for e side signals, high-ran high-range side sign information obtained by the encoding means, and generating means for generating i codestring based on signals encoded by the low-range encodi means, and on the encop In a further aspe method for decoding a for splitting a codestri encoding each of plural plural signals, high-r number of high-range plural signals and encoding the high-rang^ for decoding the sep decoding step for dec based on the encoded generating an output s side signals and the de In a further asp apparatus for decodiiji means for splitting obtained on encoding generated based on pi on encoding a number than the plural signals on encoding the high-r for decoding the se decoding means for de th ect.

ing means and the high-range encoding ing information, ct, the present invention provides a codestring including: a separating step ng having low-range codes obtained on low- range signals, generated based on ange codes obtained on encoding a signals smaller in number than the e encoded information obtained on signals, a low-range decoding step grated low-range codes, a high-range oding the separated high-range codes information, and a synthesis step for ignal based on the decoded low-range <J;oded high-range side signals.

, the present invention provides an g a codestring including separating codestring having low-range codes each of plural low-range signals,

Aral signals, high-range codes obtained of high-range signals smaller in number and the encoded information obtained ange signals, low-range decoding means parated low-range codes, high-range coding the separated high-range codes i tied e d on nt information and synthesis means for ignal based on the decoded low-range cjoded high-range side signals.

pect, the present invention provides a g recorded thereon a codestring having on encoding each of plural lowbased on plural signals, high-range oding a number of high-range signals n the plural signals and the encoded encoding the high-range signals, tion encoding method and apparatus invention, acoustic signals of plural frequency to produce a low-range side In the low-range side, signals of the signals that can be decoded on are encoded for compression. In the mber of channels smaller than in the aerated and at least the normalization thereof is encoded for compression to

6f the encoded information to enable er processing volume and a smaller edium of the present invention has codestring generated by the encoding of the present invention, thus enabling based on the encoded generating an output side signals and the de In yet another as recording medium haviij low- range codes obtai range signals, gene rat codes obtained on enc smaller in number tha information obtained

With the informa according to the prese channels are split in and a high-range side, respective channels subsequent decoding high-range side, a nu low-range side are ge coefficient information decrease the volume encoding with a small buffer memory capacity

The recording m recorded thereon the method and apparatus or storage of acoustic sig a smaller recording cap nals of higher quality with the use of acity.

According to the present invention, acoustic signals can be encoded using a smaller number of bits while the stereo sound effect or multi-channel effect, for example, is maintained. The processing volume or the buffer memory capacity required for encoding and decoding can also be reduced.

BRIEF DESCRIPTION OF THE DRAWINGS

Fig.1 is a block circuit diagram showing an illustrative ' r ding/reproducing apparatus as an reco structure of embodiment of the compressed data recording/reproducing the present invention.

ircuit diagram showing an illustrative ing circuit according to the present apparatus according to Fig.2 is a block c structure of an encod invention.

Fig.3 is a block c ircuit diagram showing an illustrative structure of a signal component encoding circuit according to the present invention.

ircuit diagram showing an illustrative structure of a transform circuit according to the present invention.

Fig.5 is a block Circuit diagram showing an illustrative structure of a decoding circuit according to the present invention.

,....,. ψ ...

Fig.6 is a block c structure of an invers present invention.

Fig.7 is a block c structure of a signal c the present invention.

Fig. 8 illustrates a

Fig.9 illustrates t encoded by the basic e

Fig. 10 shows an from frame to frame.

Fig. 1 1 shows an (L + R)/2 to a frame.

Fig. 12 illustrates components are divide|d encoding the resulting

Fig. 1 3 illustrates by the encoding meth into tonal and noise signals.

Fig.14 is a block structure of a signal for dividing the sig components and encodi

Fig.15 is a block :rcuit diagram showing an illustrative e transform circuit according to the rcuit diagram showing an illustrative omponent encoding circuit according to basic encoding method, tie structure of a codestring of a frame

r.co ding method.

example of arraying L and R channels example of arranging the channel an encoding method in which the signal into tonal and noise components and signals.

the structure of a codestring encoded od of dividing the signal components cjomponents and encoding the resulting circuit diagram showing an illustrative (Component encoding circuit configured rial components into tonal and noise ng the resulting signals.

circuit diagram showing an illustrative omponent decoding circuit configured ring obtained on dividing the signal ind noise components and encoding the structure of a signal c for decoding a codest components into tonal ί resulting signals .

Fig.16 is a block structure of an encodi channel (L + R)/2 and a c Fig.17 is a block structure of a decodii obtained on encoding tl R)/2.

Fig.18 is a block structure of an encodi)

L and R channels on tl (L + R)/2 channel on the Fig.19 is a block structure of an A-type Fig.18.

Fig.20 is a block c structure of a B-type Fig.18.

Fig.2 1 illustrates on encoding the L and and on encoding the ( side.

circuit diagram showing ng circuit configured for Channel (L-R)/2 .

a schematic encoding a circuit diagram showing a schematic g circuit for decoding a codestring e channel (L + R)/2 and the channel (Lcircuit diagram showing the schematic ng circuit configured for encoding the ie low-range side and for encoding the high-range side.

circuit diagram showing an illustrative signal component encoding circuit of ircuit diagram showing an illustrative i

signal component encoding circuit of the structure of a codestring obtained R channels on the low frequency side L + R)/2 channel on the high frequency · ..ι ·;τ

<img file="MY122169A_D0001.tif" />

Fig.22 is a block structure of an encodir

L and R channels on th the (L + R)/2 channel on Fig.23 is a block structure of an encodir

L and R channels on tl common signals for frequency side .

Fig.24 is a block structure of a signal encoding circuit of Fig Fig.25 is a flov example of the contr Fig-23 .

Fig.26 illustrates on encoding the L and and on encoding comm the high frequency side

Fig.27 is a blocl· of a decoding circuit obtained on encoding frequency side and on R channels on the high

Fig.28 is a block circuit diagram showing the schematic ig circuit configured for encoding the e low frequency side and for encoding the high frequency side.

circuit diagram showing the schematic ig circuit configured for encoding the ie low frequency side and for encoding the L and R channels on the high circuit diagram showing an illustrative component encoding circuit of the .23.

^r chart for illustrating a processing ul circuit of the encoding circuit of the structure of a codestring obtained R channels on the low frequency side on signals for the L and R channels on .

: circuit diagram showing the structure configured for decoding a code string the L and R channels on the low encoding common signals for the L and frequency side.

circuit diagram showing the schematic g circuit acco ng the L and encoding comm&lt;j) frequency side circuit diagram according to a a codestring o e low frequenc he L and R structure of an encodin configured for encodi frequency side and for R channels on the high Fig.29 is a block of a decoding circuit configured for decoding L and R channels on tl common signals for frequency side.

Fig.30 is a block structure of a decod embodiment configured encoding such that th^ side differ from each o

Fig.31 is a flowch method in which a wei circuit of Fig.30 determ Fig.32 is a block structure of an encodi the channels (L + R)/2 low-range side and channels and the wei high-range side.

Fig.33 is a flowch ng circuit acc for decoding a L and .R chan art showing the ghting decision ines the weight circuit diagram rding to an embodiment R channels on the low n signals for the L and showing the structure modified embodiment itained on encoding the y side and on encoding channels on the high circuit diagram showing a schematic ording to a modified codestring obtained on nels on the high-range :her in signal level.

processing flow of the circuit of the decoding ing parameters. showing the schematic ng circuit of aa embodiment in which and (L-R)/2 are encoded on the t|he common signals of the L and R hting parameters are encoded on the art showing the processing flow of the method in which a wei ( circuit ofFig.32 deter Fig.34 illustrates the channels (L + R)/2 ⁵ low-range side and channels and the wei high-range side.

Fig.3 5 is a block structure of an decod ¹⁰ (L + R)/2 and (L-R)/2 a the common signals weighting parameters a

Fig.36 is a fl o w ciji a weighting coefficie ¹⁵ circuit of Fig. 3 5 .

Fig.3 7 is a block schematic structure of bands embodying the p

Fig.3 8 is a bloc|&lt; ²⁰ schematic structure o bands embodying the p

DESCRIPTION OF PRE Referring to the present invention will Fig.1 shows the s ghting decision circuit of the encoding ines the weighting parameters, the structure of a codestring in case and (L-R)/2 are encoded on the the common signals of the L and R dhting parameters are encoded on the circuit diagram showing the schematic ing circuit in which the channels re encoded on the low-range side and of the L and R channels and the re encoded on the high-range side, art illustrating the processing flow of nt calculating circuit of a decoding circuit diagram for illustrating the an encoding circuit having four split resent invention.

circuit diagram for illustrating the a decoding circuit having four split resent invention.

FERRED EMBODIMENTS drawings, preferred embodiments of the be explained in detail.

chematic structure of a compressed data apparatus according to an embodiment recording/reproducing of the present invention In the compressed data recording/reproducing apparatus shown in Fig.1, a magneto-optical disc 1 run in rotation by a spindle motor (M) 51 data recording on the is used as a recording medium. During magneto-optical disc 1, a modulating magnetic field corresponding to recording data is applied by a magnetic head (H) 54 by, for example, a ma so-called magnetic fie data along a recording

The optical head such as a laser dio , as the laser light beam is illuminated gnetic head 53, by way of performing d modulation recording for recording track of the magneto-optical disc 1.

During reproduction, the recording track of the magnetooptical disc 1 is traced by the laser light by an optical head 53 for photomagnetic data reproduction.

is made up of a laser light source, de, optical components, such as a collimator lens, an objective lens, a polarization beam splitter and a cylindrical lens. This optical head 53 is mounted facing the magnetic bead 54 via magneto-optical disc 1.

on the magneto-optical disc 1, the magnetic head 54 is driven by a magnetic head driving circuit 66 of a recording system as later explained for applying the modulating magnetic field corresponding to the recording data, at the same time as the laser light beam is illuminated on a target track of the) magneto-optical disc 1 by the optical head 53 by way of per accordance with the m order to detect the foe orming thermo-magnetic recording in ignetic field modulating system. On the other hand, the optical head 53 detects the reflected light of the laser light illuminated on the target track in order to detect the focusing error by the astigmatic method and in using error by the push-pull method.

For reproducing data from the magneto-optical disc 1, the optical head 53 detect' the focusing errors and the tracking errors, at the same time as it detects the difference in the polarization angle (Kerjr rotation angle) from the target track of the laser light in ordsr to generate playback signals.

An output of the circuit 55, which extr optical head 53 is supplied to an RF acts the focusing error signals and the tracking error signals from the .output of the optical head 53 to supply the extracted signals to a servo control circuit 56, playback signals to a bi-level signal decoder 71 of the reproducing system, circuit 56 is made up of, for example, while converting the which is supplied to a The servo control a focusing servo control circuit, a tracking servo control circuit, a spindle mot servo control circuit.

:|or servo control circuit and a thread The focusing servo control circuit focusing-controls the optical system of the optical head 53 for reducing the focusing error signals to zero, while the tracking servo control circuit tracking-controls the optical system of the optical head 53 foif reducing the tracking error signals to

<img file="MY122169A_D0002.tif" />

tor servo control circuit controls the at the magneto-optical disc 1 will be zero. The spindle mo spindle motor 51 so th run in rotation at a prs-set rotational velocity, such as at a pre-set linear velocity.

The thread servo control circuit also moves the optical head 53 and the magnetic head 54 to a target track position or the magneto-optical disc 1 designated by a system controller 57. The servo control circuit 56, performing these var ious control operations, sends the information specifying the operating states of the various by the servo control circuit 56 to the ntroller 57 are connected a key input d a display unit 59. The system components controlled system controller 57.

To the system co operating unit 58 an controller 57 supervises the recording system and the reproducing system by the key input unit the operating input information from 58. The system controller 57 also supervises the recording position or the playback position on the recording track traced by the optical head 53 and the magnetic head 54, based on the sector-based address information reproduced by the header timer or subcode Q-data from the recording trabk of the magneto-optical disc 1. The system controller 57 £.lso performs control of displaying the playback time on the display unit 59 based on the data compression rate of the compressed data recording/reproducing device and the playback position information on the recording

For playback time display, the sector-based address track.

information (absolute header data or the sub of the magneto-optical time information) reproduced by the -code Q data from the recording track disc 1 is multiplied by a reciprocal of the data compression ratio, such as 4 for the 1/4 compression, in order to find the displayed on a display time information is recording track of, for pre-formatted absolute actual time information, which is unit 59. For recording, if the absolute pre-recorded (pre-formatted) on the example, a magneto-optical disc, the time information can be read out and multiplied by the reciprocal of the data compression ratio for displaying the current position in terms of the actual recording time. In this recording system the disc recording/reproducing at an input terminal 60 device, an analog audio input signal A_inis supplied via a low-pass filter (LPF) to an A/D converter 62 which then quantizes the analog audio input signal A_in.

The digital audio signal from the A/D

Converter 62 is supplied to a ATC encoder 63. The digital audio input signal D_in from the input terminal 67 is supplied via a digital input interfacing circuit (digital input) 68 to the ATC (Adaptive Transform Coding) encoder 63. The ATC encoder 63 performs bit compression (data compression) corresponding to a pie-set data compression ratio on the digital audio PCM datu of the pre-set transfer rate obtained on quantization of the input signal A_in by the A/D converter

62. The compressed data (ATC data) outputted by the pre-set data compression rat io is supplied to memory 64.

Supposing that the data compression ratio is 1/8, the data transfer rate is reduced of the CD-DA format a of 75 sectors/sec or to impossible, sector-con be explained subseque a burst fashion at the to one-eighth of the data transfer rate s the standard digital audio CD format

9.375 sectors/second.

The memory (RAM) 64 is used as a buffer memory having data write/readout controlled by the system controller 57 and which is configured for transiently holding on memory the ATC data supplied from the ATC encoder 63 for recording the data on a disc whenever the necessity arises. That is, if the data compression ratio is 1/8,,. for example, the compressed audio data supplied from the ATC encoder 63 has its data transfer rate reduced lo 1/8 of the data transfer rate for the standard CD-DA format of 75 sectors/ second, that is to 9.375 sectors/ second. It is this compressed data (ATC data) that is continuously recorded in the memory 64. For these compressed data (ATC data), it suffices to record the data at a rate of one sector per eight sectors, as discussed previously. However, since this recording every eight sectors is virtually tinuous recording is carried out, as will ntly. This recording is carried out in same data transfer rate as that for the standard CD-DA format sectors, such as 32 recording unit.

That is, the ATC (75 sectors/second), with preset plural sectors plus several sectors, as a audio data with the data compression rate of 1/8, continuously written at a low transfer rate of 9.3 75 ( = 75/8) sectors/second, are read out from the memory 64 in a burst-like mariner as recording data at the abovee of 75 sectors/second. The overall data transfer rate of (he data, thus read out and recorded, including the non-recording period, is the above-mentioned low rate of 9.3 75 sectors/second. However, the instantaneous data transfer rate within the time of the burst-like recording operation is the above-mentioned standard rate of 75 sectors/second. Therefore, if the rotational velocity of the disc is the above-mentioned standard velocity of the CD-DA format (constant linear velocity), recording is by the same recording density and the same recording pattern as those of the CD-DA format.

The ATC audio data, that is the recording data, read out from the memory 64 in the burst-like fashion at the r rate of 75 sectors/ second, is supplied n the data string supplied from the (instantaneous) transfe to an encoder 65.

memory 64 to the encoder 65, a continuous recording unit per each recording is such as 32 sectors, and a cluster made up of plural sectors, several cluster-interconnecting sectors arrayed ahead and at back of the cluster. These cluster interconnecting sectors are set so as to be longer than the interleaving length at the encoder 65, such that interleaving cannot affect data of other clusters.

The encoder 65 applies encoding for error correction, such as parity appendage and interleaving, or EFM encoding, to the recording data supplied in a burst-like fashion from the rding data encoded by the encoder 65 gnetic head driving circuit 66 To this circuit 6]6 is connected the magnetic agnetic head 54 is driven for applying memory 64. The reco

J are supplied to the ma magnetic head driving head 54 so that the m the magnetic field modulated in accordance with the recording data to the magneto-op ical disc 1.

The system controller 5 7.. performs memory control as

5 described above on the recording position for data continuously in a by this memory contro memory 64, while also controlling the continuously recording the recording

I burst-like fashion from the memory 64 on the recording track of the magnetooptical disc 1. For controlling the recording position in this position read out in a burst fashion supervised by the system controller 57 for supplying a control signal designating the recording position on the recording track of the magneto-optical disc 1 to the servo control circuit 56.

The reproducing system of the disc recording/reproducing manner, the recording from the memory 64 is device shown in Fig.4 is now explained. This reproducing system is configured for reproducing recording data continuously recorded optical disc 1 by the ab on the recording track of the magnetoove-described recording system. Thus, the reproducing system includes a decoder 71 supplied with a bi-level signal obtained by a RF circuit 55 from the playback output obtained in turn by the optical head 53 tracing the recording track of the magneto-optical disc 1 with a laser light beam. It is noted that not only the magnetooptical disc but also the read-only optical disc similar to the compact disc (CD) can The decoder 7 1 is be read.

a counterpart device of the encoder 65 of the above-described recording system. The playback output, converted into the bi-level signal by the RF circuit 55, is decoded for error correction or is EFM decoded for reproducing the ATC E.udio data having the data compression rate of 1/8 at a transfer rate of 75 sectors/ second which is faster than the normEil transfer rate. The playback data, obtained by the decoder 7 1, is supplied to a memory 72.

In the memory 72, having data write/readout controlled by the system controll the decoder 71 at the er 57, the playback data supplied from transfer rate of 75 sectors/second, is written in a burst-like fashion at the transfer rate of 75 sectors/second. In t he memory 72, the above-mentioned playback data, written at the above-mentioned transfer rate of 75 sectors/ second, is continuously read out at the transfer rate of 9.375 sectors/ second corresponding to the data compression rate of 1/8

The system controller 57 performs memory control for writing the playback data in the memory 72 at the transfer rate of 75 sectors/second, while reading out the playback data from the memory at the transfer rate of 9.375 sectors/second. The system controller 5 7, performing the memory control for controls the playback the playback data writ he memory 72 as described above, position for continuously reading out ten in the burst-like fashion from the memory 72 by the memory control from the recording track of the magneto-optical disc 1. The playback position control is !

by supervising the playback position of the playback data read fashion from the memory 72 by the system controller 57 and by supplying a control signal designating the playback position on the recording track of he magneto-optical disc 1 to the servo the optical disc 1 or t control circuit 56.

The ATC audio memory 72 at the tra supplied to an ATC decoder 73. This ATC decoder 73 is a counterpart device of system and reproduc expanding the ATC d data, continuously read out from the nsfer rate of 9.375 sectors/second, is the ATC encoder 63 of the recording es the 16-bit digital audio data by ata by a factor of eight. The digital audio data from the A converter 74.

The D/A converte supplied from the ATC

TC decoder 73 is supplied to a D/A r 74 converts the digital audio date decoder 7 3 into an analog signal for ⁵ forming an analog audio output signal A_oul. This analog audio output signal A_out, obtained from the D/A converter 74, is outputted via a low-pass filter 75 at an output terminal 76.

The high-efficiency encoding is explained in detail. Specifically, the technique of high-efficiency encoding an such as an audio PCM signal, by techniques of sub-bard coding (SBC), adaptive transform coding (ATC) and adaptive bit allocation, is explained by referring to Fig.2 ff.

In the encoding executing the method waveform signals) of waveform 110a is conv device, (encoder 63 of Fig.1) for or encoding the information (acoustic the present invention, an input signal erted by a conversion circuit 111a into signal frequency components 110b. These signal frequency components 110b are encoding circuit Illb then encoded by a signal component to produce an encoded signal 110c. A codestring generating circuit 111c then generates a codestring llOd from an encoded generating circuit 111 signal 110c generated by the codestring c .The conversion circuit 111a splits the input signal 120a

5 by a band-splitting filter 112a into two bands and resulting two band signals 120b, 120c are transformed by forward circuits 112b, 112c by MDCT into spectral signal components 120d, I20e. The input signal 120a corresponds to the signal waveform 110a of Fig.2, while the nents 120d, 120e correspond to the onents 110b shown in Fig.2. In the a, shown in Fig.3, the bandwidths of spectral signal compo signal frequency comp conversion circuit 111 the two band-split signals 120b, 120c are one-half the bandwidth of the input 120a is thinned out by the conversion circ of by MDCT. Althou frequency components signal is preferably orthogonal transform because then a large n signal 12 0a, that is, the input signal

1/2. Of course, any other structure of uit 111a may be used besides the illustrative example. For example, the input signal may be directly transformed by MDCT into spectral signals, while the input signal may also be transformed by DFT or DCT, instead gh the input signal may be split into by a band splitting filter, the input transformed by the above-mentioned methods into frequency components umber of frequency components can be obtained with a smaller volume of processing operations.

The signal comppnent encoding circuit 111b normalizes 130a from one pre-set band to another, circuit 113a, while calculating the information 130c from the signal the signal components by a normalization quantization fineness components 130a by the quantization fineness decision circuit

113b, as shown in F

g.4. The quantization circuit 113c quantizes the normalized spectral coefficient data 130b from the normalization circ fineness information 1 130a correspond to the uit 113a based on the quantization Oc. Meanwhile, the signal components encoded signal 110b of Fig.2. The encoded signal 110c of Fig.2 includes the normalization coefficient information mentioned quantization for normalization 130e and the abovefineness information 130c, in addition to the quantized signall components for normalization 130d from the quantization cjircuit 113c.

apparatus (decoder 73 of Fig.1) for signals from the codestring generated d encoder, codes 140b of the signal

In the decoding regenerating the audio bv the above-describe components, normalization coefficient information and the quantization fineness codestring 140a by a shown in Fig.5. From 140c are restored by a information are extracted from a codestring resolution circuit 114a, as these codes 140b, signal components codestring decoding circuit 114b and, from the restored signil components 140c, acoustic waveform signals 140d are regenerated by a back-conversion circuit

14c.

This back-conversion circuit 114c of the information decoder is configured as shown in Fig.6 and is associated with the conversion circuit shown in Fig.3. In the backconversion circuit 114c, shown in Fig.6, inverse orthogonal transform circuits 115a, 115b apply inverse orthogonal gnals 150a, 150b, respectively, for als, which are then synthesized by a transform to input si restoring the band sign band synthesizing filter 115c. The input signals 150a, 150b correspond to a signal

140c the signal components of which have been restored by the signal component decoding circuit

114b. An output signa 115c corresponds to t

Fig-5 .

150e of the band synthesizing filter ie acoustic waveform signal 140d of

The signal component decoding circuit 114b of Fig.5 is configured as shown in

Fig.7, and applies dequantization and

The spectral sign of the above-described for example in Fig.8. Fig.8, denote absolute denormalization processing to the codes 140b from the codestring resolution circuit 114a, that is spectral signals. In the signal component decoding circuit 114b, shown in Fig.7, a dequantization circuit 116a dequantizes input codes 160a, while the denormalization circuit 116b denormalizes the signals 160b obtained on dequantization to output signal components 160c. The above codes 160a correspond to codes 140b from the codestring resolution circuit 114a of Fig.5, while the output signal components 160c correspond to the signal components 140c of Fig.5.

Lals obtained by the conversion circuit encoder, shown in Fig.3, are as shown The spectral components, shown in values of the spectral components by herein encoding units, by forward orthogonal the encoding unit can of the band-splitting transformed directly

MDCT after level conversion by dB values. That is, in this encoder, the input signal is converted into 6 4 spectral components, from one dre-set transform block to another, and is normalized and quantized in terms of eight bands, termed shown in [1] to [8] in Fig.8. It is noted that these spectral signals transiently split into two bands by band-sρ 1 itting by the band-splitting filter, followed transform, in which the bandwidth of be set independently of the bandwidth 'liter. The input signal may also be into spectral signals by orthogonal transform without being passed through the band-splitting filter. If the quantization fineness is varied from one encoding unit to another depending on how the frequency components are distributed, there is assured encoding with perceptually high efficiency with suppression of the sound quality deterioration to the minimum.

Fig.9 shows an i lustrative structure of a codestring in case of encoding as described above.

In the present restoration of spectral block) are encoded in an illustrative structure, data for signals of each transform block (time accordance with frames each made up of a pre-set number cf bits, for a first channel for the left channel and for a sec the L and R channels ond channel as the right channel, with being arrayed alternately. In a leading end (header) of each information which is th frame are sequentially arranged the e control data such as synchronization signals and the encoded number of encoding units, encoded of bits, the information which is the information and normalization co efficient information of each encoding unit, and the spectral coefficient data, normalized and quantized on the normalization coefficient data and with a pre-set number quantization finenesb basis of the the quantization fineness data from one encoding unit to another.

sponding to the control data and the a are encoded beginning from the lowrange side encoding unit.

The number of bits actually required for decoding the spectral signals of the transform block is determined by the number of the encoded encoding units and the number of

The information corre spectral coefficient dal quantization bits spe cified by the quantization fineness information of each encoding unit. The number of bits may vary from one frame to another. The above-mentioned required number of bits, counting from the leading end of each frame, is valid during reproduction, with the remaining area of each not affecting playback signals, time block in association with a frame umber of bits, as in the present ding position of an arbitrary transform frame being a void area. By encoding each having a pre-set n embodiment, the recor block can easily be calculated when the codestring is recorded in a recording medium, such as an optical disc, for facilitating random accessing, that is reproduction from an arbitrary position. Usually, a larger number of bits are improving the sound quality for a in each frame .

lustrate an example of a recording

ρ._Π, ,Λ effectively used for minimizing the void are Figs.10 and 11 i format when chronologically recording data of the frame shows an example in which the left (L) are arrayed alternately from frame to frame, and Fig.11 shows signals having sample values produced on (L + R)/2-ing L and R channels from frame to frame. The channel obtained in this manner on (L + R)/2-ing L and R channels from frame to frame is termed herein a (L + R) channel. Similarly, the channel obtained in this manner on (L-R)/2-ing L and R channels from frame to frame is termed herein a (L-R) channel.

shown in Fig.9. Fig.10 and right (R) channels

By employing the the two channels of L recording format as shown in Fig.10, and R can be recorded on the same recording medium. If the recording format as shown in Fig.11 is used, in which tw alternately from frame recorded/reproduced, without complicating t Although the tech o channels of L and R are arrayed to frame, double time signals can be while reproduction can be realized ie reproducing circuit.

nique explained with reference to Fig.9 hove, the encoding efficiency can be has been discussed a ' I further improved over the encoding method of Fig.9.

For example, the encoding efficiency can be improved by employing the so-called variable encoding technique of allocating shorter and longer codelengths for the signals of higher and lower probability of occurrence, respectively.

Also, if the above-mentioned pre-set transform block in encoding input signals, that is the time block length for orthogonal transform, is longer, the quantity of the subsidiary information, such as the quantization fineness information or the normalization coefficient information, can be reduced per block, while the freque finely, thus improving ncy resolution can be controlled more he encoding efficiency.

In addition, if a niethod disclosed in PCT Application of the International Pub ication . WO94/28633 by the present

Assignee, that is a method of separating perceptually crucial tonal components, that is concentrated in a s signals components, is signal components where the energy pecified frequency, from the spectral and encoding the separated tonal components independently of the remaining spectral signal components, efficient encoding may be realized with a high compression ratio without substantially producing perceptual deterioration of audio Referring to Fig.

signals.

12, the method of separating the tonal signal components and encoding the separated signal components is explain ed. In the example of Fig.12, a set of three tonal component; have been separated as tonal signal components from the s pectral signal components. The signal components making up each tonal component are encoded along with respective the tonal components.

In general, the components, where the small number of spectr extremely high pre deterioration. Howev encoding unit freed of position data on the frequency axis of signal components of the tonal energy is concentrated in a relatively al components, need to be quantized to cision for evading sound quality er, the spectral coefficients in each the tonal components can be quantized with a relatively small number of bits without deteriorating the perceptual sound quality.

Although only a relatively small number of spectral signal components are; shown in Fig.12 for simplifying the drawing, the signal elnergy is concentrated in a few signal components of tens of signal components making up a given encoding unit. Therefore, the amount of data is not increased result of separation of the tonal at the encoding efficiency can be by separating the tonal components.

significantly as a components, such th improved on the whole

Fig.13 shows an illustrative structure of a codestring in case of encoding by t Fig.12. In the present at the leading end of he method explained with reference to illustrative structure, there is arrayed, each frame, the information comprised of control data, such as sync signals and the encoded number tonal components on of the encoding units, encoded with a pre-set number of bits, as a header portion. Next to the header portion is arrayed the information comprised of the encoded tonal component data as tonal component data.

As tonal component data, the information which is the encoded number of signal components in the tonal components is arrayed first, and is followed by the information which is the encoded position information of the the frequency axis, the information which is the encoded information on the normalization coefficients and the information which is the normalized, tonal signal components, in this order.

Next to the tonal component data is arrayed the information which is the encoded data of the residual signal quantized and encoded left after subtraction of the tonal signal components from the original spectral signal components. This residual signal can also be termed noisy signal components. This residual signal zation fineness data and normalization Jch encoding unit, spectral component quantized based on the normalization is comprised of quanti coefficient data of ea signals normalized and coefficients data and components other than the quantization fineness data (signal the tonal components) encoded in the order of the increasing frequency of the encoding units. It is noted that spectral signal components of the tonal and other signal components (coe Fig. 14 shows an component encoding cii 'ficient data) are encoded by VLC. illustrative example of the signal cuit 111b of Fig.2 when separating the tonal signal components from the above-mentioned respective signal components.

In the signal component encoding circuit 111b, shown in Fig.14, the signal components 170a (110b) sent from the conversion circuit 111a of Fig.2 are sent to a tonal component separation circuit 117a. The signal components 170a are *nal components 170b and other signal -tonal signal components). The tonal 0b are sent to a tonal component while the non-tonal signal components onal component encoding circuit 117c.

separated into tonal si components 170c (non signal components 17 encoding circuit 117b, 170c are sent to a nonThe tonal component encoding circuit 117b and the non-tonal component encoding circuit 117c encode the signal constituting the tonal components supplied thereto to output resulting output signals 170d, 170e. The tonal component encoding circuit 117b generates the information constituting the tonal component data of Fig.13, at the pme time as it encodes the information component data of Fig.13. The signal encoding arrangements in the tonal component encoding circuit 117b and in the non-tonal component encoding circuit 117c are the same as those shown in Fig.4.

Fig.15 shows an illustrative example of the signal component decoding circuit 114b in case the tonal signal components have been components .

In the signal comp Fig.15, the code 140a s circuit 114a of Fig.5 is separated from the respective signal tonal component deco tonal component deco tonal signal compone components 180d are b circuit 118c, which onent decoding circuit 114b, shown in upplied from the codestring resolution made up of tonal component data 180a and non-tonal component data 180b, which are sent to associated tonal component decoding circuit 118a and nonding circuit 118b, respectively. The ding circuit 118a decodes the tonal signal components from the tonal component data shown in Fig.13 to output resulting tonal signal components 180c. The non-tonal component decoding circuit 118b decodes the tonal signal components from the non-tonal component data to output resulting non-tjonal signal components 180d. These nts 180c and the non-tonal signal oth sent to a spectral signal synthesis then synthesizes the tonal signal components and the nob-tonal signal components based on the above-mentioned position data to output the resulting signal decoding configurations of the tonal circuit 118a and the non-tonal rcuit 118b are the same as those shown components 180e. The component decoding component decoding cii in Fig.7.

Although the method for efficient encoding of signals of respective channels has been explained above, there is also known a method for further improving the encoding efficiency correlation between channels. For L -channel signals are substantially the R-channel signals, signals having (L + R)/2 and those having the sample by exploiting signal example, if, when the similar in waveform to the sample values of values of (L-R)/2 are encoded in place of encoding the L and

R channels, the signal encoding can be done w Fig.16 shows an device for encoding by Referring to Fig.

The signal compo 119b and the signal circuit 119c are sen (L-R)/2 is of a smaller value, so that ith a smaller number of bits, illustrative structure of an encoding the above-mentioned method.

6, a channel conversion circuit 119a converts a L-channel signal 190a and a R-channel signal 190b into a (L + R) channel signal 190c (signal having sampled value of (L + R)/2) and a (L-R) channel signal 190d (signal having sampled value of (L-R)/2).

The signal 190c of the (L + R) channel and the signal 190d of the (L-R) channel are sent to the conversion circuits 119b, 119c, respectively, so as to be converted as in the conversion circuit 11 la of Fig.2.

nents 190e from the conversion circuit components 190f from the conversion t to the signal component encoding circuits 119e and 1 1 9f, respectively. These signal component encoding circuits ll9e and 1 19f operate similarly to the signal component encoding circuit 111b of Fig.2. The signal components 190e, 190f are also sent to the control circuit

19d.

The control circuit 1 19d determines the number of allocated bits 190g in encoding the signal components of the (L + R) channel in the signal component encoding circuit 119e and the number of allocated bits 190h in encoding the signal components of the (L-R) channel in the signal component , respectively, based on the signal of the conversion circuits 119b, 119c, encoding circuit 119 components 190e, 190f

190f for the (L-R) cha bits 190g, 190h as dete respectively. During this bit number decision, the entire bits can also be allocated so that the number of bits will be proportionate to the signals energies in each channel.

Thus, the signal components 190e, 190f encode the signal components 190e for the (L + R) channel the signal components tnnel based on the numbers of allocated trmined by the control circuit 1 1 9d.

The encoding by the signal component encoding circuits

119e, 1 19f generates codes 190i, 190j both of which are sent i

to a codestring generating circuit 119g. This codestring generating circuit ll9g generates and outputs a codestring 190k from the codes 190i, 190j of the (L + R) and (L-R) channels, respectively:

Fig.17 shows ah illustrative structure of a decoding circuit adapted for decoding a codestring 190k generated by

I ⁴³ j

the encoding device shown in Fig.16.

In Fig.17, a codestring separation circuit 211a separates a signal 210b of the (L + R) channel and a signal of the (L-R) channel 210c from the codestring 210a which is the abovementioned codestring 190k.

The code 210b of the (L + R) channel and the signal 210c of the (L-R) channel are sent to signal component decoding circuits 211b and 211c, respectively. These signal component decoding circuits 211b and 211c decode the codes similarly to the signal component decoding circuit 114b of Fig.5.

The signal component of the (L + R.) channel and the signal component of the (L-R) channel, obtained by decoding by the signal component decoding circuits 211b and 211c, are sent to associated inverse transform circuits 2 1 ld, 211e, respectively. The inverse transform circuits 21 ld, 211e perform inverse transform similarly to the inverse transform i

circuit 114c shown in F|ig.5.

I

The signal 210f of the (L + R) channel, obtained by the inverse transform by t the signal 210g of the :he inverse transform circuit 21 ld, and (L-R.) channel, obtained by the inverse transform by the inverse transform circuit 211e, are both sent to a channel conversion circuit 21 1 f, which converts the (L + R) channel signals and the (L-R) channel signals into a Lchannel signal 210h a outputted.

nd a R-channel signal 210i, which are ' Τ-f44

In addition to the above-described method, there is also known a method for efficiently encoding the L and R channel signals by exploiting characteristics of the human hearing system. This method realizes efficient encoding by exploiting the fact that the phase difference of the L and R channel signals contributes to the psychoacoustic stereo effect mainly in case the signals are low-range signals. Specifically, the signals of both the L and R channels are encoded on the low frequency side and, on the high-range side, the signal waveform of the (L + R) channel is normalized and quantized using the normalization coefficients different for the L and R channels.

Fig.18 shows an illustrative structure of an encoding device employing this method for encoding.

To the encoding device, shown in Fig.18, a L-channel signal 220a and a R-channel signal 220b are entered and transformed by associated transform circuits 221a, 221b, respectively, for transform as by the transform circuits 221a, 221b, respectively. These transform circuits 221a, 221b output signal components of the respective channels 220c, 220d which are sent to the associated signal component encoding circuits 220f, 220g. The signal component encoding circuits 221f, 221g are hereinafter referred to as B-type signal component encoding circuits. These B-type signal component encoding circuits 221f, 221g encode low-range signal components of the L-channel signal components 220c and the

R-channel signal components 220d, respectively.

The signal components of the respective channels 220c, 220d from the associated transform circuits 221a, 221b are also sent to a channel synthesis circuit 221d which sums the L-channel signal components 220c and the R-channel signal components 220d together to produce (L + R) channel signal components 2 2 Oh. These (L + R) channel signal components 220h are sent to a signal components encoding circuit 221e. The signal component encoding circuit 220e is hereinafter referred to as an A-type signal component encoding circuit.

This A-type signal component encoding circuit 220e normalizes and quantizes the high-range side signal components of the (L + R) channel signal components as described above using the normalization coefficients different for the L and R channels.

The control circuit 221c is substantially similar to the control circuit 1 19d of Fig.16. The control circuit 211c of Fig.18 determines the allocated number of bits for encoding 220e for the (L + R) channel from the channel synthesis circuit 211d, allocated number of bits for encoding 220f for the L channel signal components 220c and the allocated number of bits for encoding 22Og for the R channel signal components 220d.

Therefore, the A-type signal component encoding circuit

221e and the B-type signal component encoding circuits 221f,

221g encode the (L + R) channel signal components 220h, Lchannel signal components 220c and the R-channel signal components 220d based on the allocated numbers of bits 220e,

220f and 220g determined by the control circuit 221c.

The encoding by the A-type signal component encoding circuit 221e and the B-type signal component encoding circuits 221f, 221g results in formation of codes 220k, 2201 and 220m which are sent to a codestring generating circuit 221h. The codestring generating circuit then generates a codestring 220n from the codes 220k, 2201 and 220m to output the generated codestring 220n.

Fig.19 shows an illustrative structure of the A-type signal component encoding circuit 221e in the configuration of Fig.18. The signal component encoding circuit 221e of Fig.19 basically is of the structure similar to the structure of the signal component encoding circuit shown in Fig.4, with the difference being that the output signal of the signal component encoding circuit 221e is devoid of the normalization coefficient information.

In Fig.19, the signal component 230a, which is the signal component 220h of the (L + R) channel from the channel synthesis circuit 221d, is normalized from one pre-set band to another by a normalization circuit 231a and sent to a quantization fineness decision circuit 231b. The quantization fineness decision circuit 231b calculates the quantization fineness information 230e based on the above-mentioned signal components 230a and the number of allocated bits 230b corresponding to the above-mentioned number of allocated bits

220e.

The normalized spectral coefficient data 230c from the normalization circuit 231a and the quantization fineness information 230e from the quantization fineness decision circuit 231b are sent to a quantization circuit 231c which then quantizes the normalized spectral coefficient data 230c based on the quantization fineness information 230e. The quantization by the quantization circuit 231c gives codes 230f which are outputted as codes 220k of Fig.18 simultaneously with the quantization fineness information 230e.

Fig. 2 0 shows an illustrative structure of the B-type signal component encoding circuits 221f, 221g.

In Fig.20, the B-type signal component encoding circuits separate signal components 24 0a, which are the L-channel signal 220c from the transform circuit 221a of Fig.18 or the R-channel signal components 220d from the transform circuit, into low-range signal components 240c and high-range signal components 240d, by a signal separation circuit 241a.

The low-range signal components 240c are encoded by the normalization circuit 241b, quantization circuit 241e and the quantization fineness decision circuit 241d. The ί

quantization fineness decision circuit 241d determines the quantization fineness based on the number of allocated bits for encoding 240b from the control circuit 221c of Fig.18.

On the other hand, the high-range side signal components 240d are normalized by the normalization circuit 241c so that only normalized spectral coefficient data are outputted.

The quantization fineness information 240f from the lowrange side quantization fineness decision circuit 241d, codes 240h from the quantization circuit 241e, normalization coefficient information 240i from the normalization circuit 241b and the and the normalized spectral coefficient data 240g from the high-range side normalization circuit 241c are sent as codes 2201 or 220m of Fig.18 to a codestring generating circuit 221h of Fig. 18.

Fig.21 shows an illustrative structure of a codestring generated by the codestring generating circuit 221h of Fig.18.

In Fig.21, the codestring is made up of a header composed of synchronization signals and the number of encoding units for the low frequency range, encoded data of the low-range side of the first channel (L-channel), encoded data of the low-range side of the second channel (R-channel) and encoded data of the high-range side. As for the low-range side, quantization fineness data, normalization coefficient data and spectral coefficient data are given as independent encoded data for the first channel (L-channel) and for the

4'Λ·γ'.4· second channel (R-channel). However, as for the high-range side, data common to the two channels (first and second channels) are given except the normalization coefficient information for the first channel (L-channel) and the second channel (R-channel). Thus, in Fig.21, the quantization fineness information and the spectral coefficient data are common data for the first and second channels.

Fig.22 shows an illustrative structure of a decoding device configured for decoding encoded data of the codestring shown in Fig.2 1.

In Fig.22, encoded data 250a of the codestring shown in Fig.21 is separated by a codestring separation circuit 251a into L- and R-channels. However, high-range data, which are common data for the two channels, are sent to both the signal component decoding circuits 251b and 251c. The L-channel encoded data 250b are decoded by a signal component decoding circuit 25 1b, while the R-channel encoded data 250c are decoded by a signal component decoding circuit 251c.

The signal components 250d, 250e, decoded by these signal component decoding circuits 251b, 251c, are sent to associated inverse transform circuits 251d, 251e where the signal components are inverse-transformed for restoration of the L-channel time-domain signals 25Of and the R-channel time-domain signals.

In the method for encoding the channel corresponding to the (L + R) channel and to the (L-R) channel for the entire range as explained with reference to Fig.16, these two channels need to be encoded even if one of the channels is of a low signal level. This imposes a limit in improving the compression efficiency by encoding. With the method of encoding only the normalization coefficients for the high range side, as explained in Fig.18, a corresponding number of bits are required for encoding the normalization coefficients of the entire range. Also, with the above-described method, transform processing and inverse transform processing need to be performed for the entire frequency range for both the L and R channels, thus requiring voluminous processing and buffer memory space.

Thus, in the present embodiment, both the L and R channel signals or signals capable of restoring the L and R signals are encoded for the low range side. On the other hand, as for the high-range signals, common signals are used for the L and R channels, or the common signals for the L and R channels modified only in signal level are adapted to be reproduced for realization of highly efficient encoding as well as for reducing the volume of the processing for the decoding and encoding. This will be explained with reference to the drawings.

Fig.23 shows an illustrative structure of an encoding device adapted for performing the encoding by the method of the present embodiment described above.

In Fig.23, the L-channel signal 260a and the R-channel signal 260b are supplied to transform circuits 261a, 261b, respectively. These transform circuits 261a, 261b are similar to those described previously and perform transform processing for the respective input signals. The L-channel signal components 260c, transformed by the transform circuit 261a, are sent to a low-range signal separation circuit 261g, while the R-channel signal components 260d, transformed by the transform circuit 261b, are sent to a low-range signal separation circuit 261h.

The low-range signal separation circuits 261g, 261h separate only the signal components of the low-range side of the supplied signal components to transmit the separated lowrange signals as low-range signal components 260j, 260k to associated signal component encoding circuits 26 lj, 261k.

The signal component encoding circuit 26 lj, fed with the low-range side L-channel signal components 260j, and the signal component encoding circuit 261k, fed with the lowrange side R-channel signal components 260k, encode lowrange signal components of the L and R channels. The signal component encoding circuits 26 lj, 261k output codes 260m, 260n which are sent to a codestring generating circuit 2611.

Although the low-range signal components of the L and R channels are encoded, it is possible to encode signals

<img file="MY122169A_D0003.tif" />

capable of restoring low-range signal components of both the

L and R channels.

The signal components of the respective channels 260c, 260d from the transform circuits 261a, 261b are both sent to a channel synthesis circuit 261e, which then sums the signal components 260c, 260d of the L and R channels to output the resulting (L + R) channel signal components 260h. The (L + R) channel signal components 260h are sent to a high-range signal separating circuit 261f.

The high-range signal separating circuit 261f separates only the high-range side signal components of the supplied (L + R) channel signal components 260h to produce high-range signal components 260i which are sent to a signal component encoding circuit 26 li.

The signal component encoding circuit 26 li encodes the (L + R) channel high-range signal components 260i to produce high-range codes 260i which are sent to a codestring generating circuit 2611.

Although the (L + R) channel high-range signal components, which are signals common to L and R channels, are encoded in the present embodiment, it is also possible to encode signals common to the L and R channels and which are modified only as to the signal level.

The control circuit 261c determines the allocated number of bits for encoding 2 60e for the high-range signal ι ....

components 260i for the (L+R) channel, allocated number of bits for encoding 260f for the low-range signal components 260j for the L-channel and allocated number of bits for encoding 260g for the low-range signal components 260k for the R-channel. The particular method for controlling the allocated number of bits for encoding by the control circuit 261c will be explained later on specifically.

Therefore, the signal component encoding circuits 261i, 26 lj and 261k encode the high-range signal components 260i for the (L + R) channel, low-range signal components 260j for the L-channel and the low-range signal components 2 6 0k for the R-channel, respectively, based on the allocated numbers of bits 260e, 260f and 260g as determined by the abovementioned control circuit 261c.

The above-mentioned codestring generating circuit 2611 generates a codestring 260o from the codes 2601, 260m and 260n supplied from the signal component encoding circuits 261i, 126 lj and 261k, respectively.

Fig.24 shows an illustrative structure of the signal components encoding circuits 26 li, 26 lj and 261k of Fig.23.

In Fig.24, a signal component 270a, which is one of the high-range signal components 260i for the (L + R) channel, low-range signal components 260j for the L-channel and the low-range signal components 2 6 0k for the R-channel, is normalized from one pre-set band to another by a normalization circuit 271a, while being sent to a quantization ₍ fineness decision circuit 271b. This quantization fineness decision circuit 271b calculates the quantization fineness information 270d, based on the signal components 270a and the signal component 270b which corresponds to one of the allocated numbers of bits 260e, 260f and 260g for encoding.

The normalized spectral coefficient data 270c from the normalization circuit 271a and the quantization fineness information 270d from the quantization fineness decision ¹⁰ circuit 271b are sent to a quantization circuit 271c. The quantization circuit 271c quantizes the normalized spectral coefficient data 270c based on the quantization fineness information 270d. The quantized codes 270e from the quantization circuit 271c are outputted as one of the codes ¹⁵ 2601, 260m and 260n simultaneously with the above-mentioned quantization fineness information 270d and normalization coefficient information 270f.

Fig.25 shows a processing example of finding data on the number of usable bits allocated by the control circuit 261c of ²⁰ Fig.23 to the signal component encoding circuits 26 li, 26 lj and 261k in each frame. This number of usable bits is that specified by the above-mentioned numbers of allocated bits 260e, 260f and 260g. In Fig.25, the data on the number of frame-based usable bits for the high-range signal components ²⁵ 260i of the (L + R) channel is specified as Bh, whereas the data on the number of frame-based usable bits for the low1 range signal components 260j of the L channel is specified as Bl and the data on the number of frame-based usable bits for the low-range signal components 260k of the R channel is ⁵ specified as B2. That is, the number of bits specified by the data on the number of bits Bh corresponds to the frame-based number of bits of the code 2601 outputted by the signal component encoding circuit 261i of Fig.23, whereas the number of bits specified by the data on the number of bits Bl ¹⁰ corresponds to the frame-based number of bits of the code 260m outputted by the signal component encoding circuit 26 lj of Fig.23 and the number of bits specified by the data on the number of bits B2 corresponds to the frame-based number of bits of the code 260n outputted by the signal component ¹⁵ encoding circuit 261k of Fig.23.

Referring to Fig.25, the high-range side signal energy Eh of the (L + R) channel is found at step S101. The signal energy El of the low-range side signal of the L-channel is found at step S102, and the energy E2 of the low-range side signal of ²⁰ the R-channel is found at step S103.

At step S104, signal energies Eh, El and E2, thus found, are processed with weighted addition using a weighting coefficient Ah for the (L+R) channel, a weighting coefficient

Al for the L-channel and a weighting coefficient A2 for the

5

R-channel, in order to find the results of processing S.

At steps S105 and S106, the total number of bits B that can be allocated to other than the header portion for the L and R channels is distributed as the above-mentioned data for the numbers of bits Bl and B2 so as to be proportionate to ⁵ the weighting energy of the respective encoding portions of the signal component encoding circuits 261j, 261k. At step S107, the remaining portions of the total number of bits is allocated to the encoding portion by the signal component encoding circuit 261i as the above-mentioned data on the ¹⁰ number of bits Bh.

Fig. 2 6 shows an illustrative structure of a code string outputted by the encoding device of fig.23.

In Fig.26, the code string is made up of a header, composed of synchronization signals and the number of low¹⁵ range encoding units, low-range side encoded data of the first channel (L-channel) (quantization fineness information, normalization coefficient information and spectral coefficients data), low-range side encoded data of the second channel (Rchannel) (quantization fineness information, normalization ²⁰ co efficient information and spectral co efficients data) and low-range side encoded data of the (L + R) channel (quantization fineness information, normalization coefficient information and spectral coefficients data). In Fig.26, since one channel of the high-range side normalization co efficient

5 information suffices in contradistinction from the codestring of Fig.21, encoding can be done with a smaller number of bits. If the high-range side signals are used in common, the spread-out feeling of the sound is slightly deteriorated. However, the user can still enjoy stereo sound reproduction because it is the low-range side signals, rather than the highrange side signals, that are more instrumental in increasing the stereophonic sound feeling.

Fig.27 shows an illustrative structure of a decoding device for decoding the codestring shown in Fig.26.

In Fig.27, encoded data 280a of the codestring shown in Fig.26 is separated by a codestring separating circuit 281a into the high-range side (L + R) channel and the low-range side R-channel. Encoded data 280b of the high-range side (L + R) channel are decoded by a signal component decoding circuit 281b, while encoded data 280c of the low-range side Lchannel are decoded by a signal component decoding circuit 281c and encoded data 280d of the low-range side R-channel are decoded by a signal component decoding circuit 281d.

The signal components 280e, 280f of the high-range side (L + R) channel and the low-range side L-channel, decoded by the signal components decoding circuits 281b and 281c, respectively, are sent to a signal component synthesis circuit 281e. The signal components 280e, 280f of the high-range side (L + R) channel and the low-range side R-channel, decoded by the signal component decoding circuits 281b and 28 ld, respectively, are sent to a signal component synthesis circuit

If.

The signal component synthesis circuit 281esynthesizes the signal components 280e of the high-range side (L + R) channel and the signal components 280f of the lew-range side L-channel to synthesize low-range and high-rage sides to send the L-channel signal components 280h of tleentire range resulting from the synthesis to an inverse transform circuit 28 lh.

The signal component synthesis circuit 28lfsynthesizes the signal components 280e of the high-range side (L + R) channel and the signal components 280g of the lw-range side R-channel to synthesize low-range and high-rage sides to send the L-channel signal components 280i of tieentire range resulting from the synthesis to an inverse transform circuit 28 li.

These inverse transform circuits 281h, 211 i inversetransform signal components of the L and R channels of the entire range to restore time -domain signals 2Hj of the Lchannel and time-domain signals 280k of the R-dbannel.

As will be apparent from the foregoing description, stereo signals can be encoded with a smaller amber of bits by the method embodying the present invention.

Also, the processing volume and the baffer memory volume can be decreased effectively by exploiting the method according to a modified embodiment of the present invention.

This modification for decreasing the processing volume and the buffer memory volume is hereinafter explained.

Fig.28 shows an illustrative structure of an encoding device of the modification in which processing is by a bandsplitting filter and a forward orthogonal transform circuit used in combination.

In Fig.28, a L-channel signal 290a and a R-channel signal 290b are sent to band-splitting filters 291a, 291b, respectively. The band-splitting filter 291a splits the Lchannel signal 2 90a into low-range and high-range signals, while the band-splitting filter 291b splits the R-channel signal 290b into low-range and high-range signals.

The low-range signal of the L-channel 290d, split by the band-splitting filter 291a, is sent to a forward orthogonal transform circuit 291e, while the low-range signal of the Rchannel 290f, split by the band-splitting filter 291b, is sent to a forward orthogonal transform circuit 290j.

The L and R channel high-range signals 290c, 290e, obtained on splitting by the band-splitting filters 291a, 291b, are sent to a channel synthesis circuit 291c so as to be synthesized to a high-range signal of the (L + R) channel which is sent to a forward orthogonal transform circuit 291d.

These forward orthogonal transform circuit 291d, 291e, 291f process the input signals with forward orthogonal ,1.

transform to produce signal components 290h, 290i, 290j which are sent to associated signal component encoding circuits 291h, 29 li, 29 lj.

The signal components 290h, 290i, 290j from the forward ⁵ orthogonal transform circuits 291d, 291e, 291f are also sent to a control circuit 291g. The control circuit 291g determines the allocated number of bits for encoding 290k for the highrange signal components 290h of the (L + R) channel, allocated number of bits for encoding 2901 for the low-range signal ⁰ components 290i of the L channel and the allocated number of bits for encoding 2.9 0m for the low-range signal components 290j of the R channel, as in the case of Fig.23, based on the signal components 290h, 290i and 290j from the forward orthogonal transform circuits 291d, 291e and 290f.

⁵ Thus, the above-mentioned signal component encoding circuits 291h, 29 li, 29 lj encode the associated high-range signal components 290h of the (L + R) channel 290h, low-range signal components 2 9 0i of the L-channel and the low-range signal components 290j of the R-channel, based on the &gt; Π allocated numbers of bits 290k, 2901 and 290m as set by the control circuit 291g.

The codestring generating circuit 291k generates a codestring 290q, similar to that shown in Fig.26, from the codes 290n, 290o and 2 90p from the signal component &gt;5 encoding circuits 29 lh, 29 li, 291j, and outputs the generated codestring 290j .

Noteworthy with the structure of Fig.28 is that, since the forward orthogonal transform on the high range side is performed in common on both channels ((L + R) channel), processing such as encoding needs to be performed only once, thus saving the processing volume and the buffer memory space. Specifically, with the above-described method of outputting the codestring shown in Fig.21, the normalization coefficient data need to be found independently for the L and R channels, even although the combination of the bandsplitting filter and the forward orthogonal transform circuits is used as the transform circuit, so that processing needs to be performed on two channels inclusive of transform on the high-range side. With the method applied to the structure of Fig.28, only one forward orthogonal transform processing operation for high-range side signals suffices for two channels.

Fig.29 shows an illustrative structure of a decoding device adapted for decoding a codestring generated by the encoding device shown in Fig.28.

In Fig.29, encoded data 310a of a codestring similar to one shown in Fig.26 is split by a codestring dividing circuit 311a into a high-range side (L + R) channel and low-range side L and R channels. Encoded data of the high-range side (L + R) channel 310b is decoded by a signal component decoding circuit 311b, while encoded data of the low-range side Lchannel 310c is decoded by a signal component decoding circuit 311c and encoded data of the low-range side R-channel

310d is decoded by a signal component decoding circuit 3 1 ld.

The signal components 310e, 310f, 310g of the highrange side (L + R) channel, low-range side L-channel and the low-range side R-channel, decoded by the signal component decoding circuits 311b, 311c and 3 1 ld, are sent to associated inverse orthogonal transform circuits 311e, 3 1 If and 3 1 lg for inverse orthogonal transform.

The signal of the high-range side (L + R) channel 310h from the inverse orthogonal transform circuit 3 1 le and the signal of the low-range side R-channel 3 lOi from the inverse orthogonal transform circuit 31 If are sent to a band-synthesis filter 3 1 lh, which then synthesizes the high-range side (L + R) channel 310h and the low-range side R-channel 3 lOi for synthesizing the low and high ranges to output the resulting full-range L-channel time-domain signals 3 10k.

The signal of the high-range side (L + R) channel 310h from the inverse orthogonal transform circuit 3 1 le and the signal of the low-range side R-channel 3 1 Oj from the inverse orthogonal transform circuit 3 1 lg are sent to a band-synthesis filter 3 1 li, which then synthesizes the high-range side (L + R) channel 310h and the low-range side R-channel 3 1 Oj for synthesizing the low and high ranges to output the resulting full-range R-channel time-domain signals 3101.

It is seen that, with the decoding device shown in Fig.29, it suffices to perform the high-range side inverse orthogonal transform only once as the common (L + R) channel. With the method for decoding the codestring shown in Fig.22, it is necessary to carry out the inverse orthogonal transform processing independently for the L and R channels for the entire frequency range, even if the combination of the inverse orthogonal transform and the band-synthesis filter is used as described above as the inverse transform circuit, because the normalization coefficients differ from channel to channel. With the above-described decoding method according to the present embodiment, it is possible to save the processing volume and the buffer memory space.

The codestring shown in Fig.26 can be reproduced so that the high-range side L and R channels will be of a different signal level for improving the stereo sound feeling.

Fig.30 shows the structure of such decoding device.

In Fig.30, encoded data 320a of a codestring similar to that shown in Fig.26 is split by the codestring dividing circuit 321a into a high-range side (L + R) channel and into a low-range side R-channel. The encoded data 320b of the highrange side (L + R) channel is decoded by a signal component decoding circuit 321b, while the encoded data 320c of the low-range side L channel is decoded by a signal component decoding circuit 321c and the encoded data 320d of the low) range side R channel is decoded by a signal component decoding circuit 321d.

The signal components 320e, 320f and 320g of the high⁵ range side (L + R) channel, low-range side L channel and the low-range side R channel are sent to associated inverse orthogonal transform circuits 321e, 3 2 If and 321g for inverse orthogonal transform.

The L-channel low-range signal 320i from the inverse ¹⁰ orthogonal transform circuit 321f and the low-range side R channel signals 3 20j from the inverse orthogonal transform circuit 321g are sent to associated band-synthesis filters 321k and 3211 and simultaneously to a weighting decision circuit 321h. This weighting decision circuit 321h sets weighting ¹⁵ coefficients for the L and R channels to send the weighting coefficients for the L and R channels thus set to associated weighting circuits 32 li, 32 lj.

These weighting circuits 32 li, 32 lj are fed with highrange signals 320h of the (L + R) channel from the inverse

Π orthogonal transform circuit 321e. These weighting circuits 3 2 li, 3 21 j perform weighting independently for the L and R channels on the (L + R) channel high-range signal 320h. The (L + R) channel high-range signals 320m, weighted for the Lchannel by the weighting circuit 32 li, is sent to the band²⁵ synthesis filter 321k, while the (L + R) channel high-range ,&lt; .ρ·_ signals 320n, weighted for the R-channel by the weighting circuit 32 lj, is sent to the band-synthesis filter 3211.

That is, in the decoding device of Fig.30, the weighting decision circuit 321h determines the weighting coefficients 320k, 3201 from the low-range side signal levels of the L and R channels, while the weighting circuits 321i, 321 j correct the sample values of the respective signals so as to be proportional to the weighting coefficients 320k, 3201.

The band-synthesis filter 321k synthesizes the high-range signals for the (L + R) channel 320m, weighted for the Lchannel, and the above-mentioned low-range signals 320i of the L-channel, by way of synthesis of the low and high ranges, to output the resulting L-channel time-domain signal 320o of the entire range.

The band-synthesis filter 3211 synthesizes the high-range signals for the (L + R) channel 320n, weighted for the Rchannel, and the above-mentioned low-range signals of the Rchannel, by way of synthesis of the low and high ranges, to output the resulting R-channel time-domain signal 320p of the entire range.

Fig.31 shows a processing example in which the weighting decision circuit 321h of the decoding device of Fig.30 determines the weighting coefficient 320k for the first channel (L-channel) (indicated as a weighting parameter rl in Fig.31) and the weighting coefficient 3201 for the second channel (R-channel) (indicated as a weighting parameter r2 in

Fig.31).

In this processing, the high-range side signal energy is also distributed to the L and R channels so as to be proportionate to the signal energy on the plow-range side. To this end, at step S201, the high-range side signal energy Ell of the L-channel is found, using the L-channel low-range signals 320i, so that the signal energy Ell will be proportionate to the energy of the L-channel low-range signals 320i. Similarly, at step S202, the high-range side signal energy El2 of the R-channel is found, using the R-channel low-range signals 320j, so that the signal energy EI2 will be proportionate to the energy of the R-channel low-range signals 3 20j .

Then, at step S203, the sum SI of the high-range side signal energy Ell for the L-channel and the high-range side signal energy EI2 for the R-channel is found. At step S204, a square root of the quotient of the signal energy Ell by the sum SI is adopted as weighting parameter rl for the first channel (L-channel). Similarly, a square root of the quotient of the signal energy EI2 by the sum SI is adopted as weighting parameter r2 for the second channel (R-channel). As will be readily understood from the relation between the weighting parameters rl and r2, the weighting parameter r2 can be found from the equation of step S205.

In the above example, the high-range side signals are used in common for the L and R channels. However, levelcontrolling data for the L and R channels, such as weighting parameters, may also be included in the codestring. This ⁵ gives rise to more faithful stereo sound feeling. The lowrange signal may also be encoded after transform into (L + R) channel and (L-R) channel instead of being encoded as L and R channels. By so doing, the (L-R) channel signal is lowered in level to realize more efficient encoding especially if these ¹⁰ is a strong correlation between the L and R channels.

Fig.32 shows an illustrative structure of a modification of the encoding device comprised of the above-mentioned two methods.

In Fig.32, a L-channel signal 33 0a and a R-channel ¹⁵ signal 330b are fed to band-splitting filters 331a and 331b, respectively. The band-splitting filter 331a splits the Lchannel signal 3 30a into low-range and high-range signals, whereas the band-splitting filter 331b splits the R-channel signal 330b into low-range and high-range signals.

0

The L-channel low-range signals 330d, split by the bandsplitting filter 331a, and the R-channel low-range signals 3 30f, split by the band-splitting filter 331b, are both sent to the channel conversion circuit 331d.

The channel conversion circuit 331d converts the L25 channel low-range signals 3 3 0d and the R-channel low-range signals 3 30f into a (L + R) channel low-range signal 330h and

I a (L-R) channel low-range signal 330i.

The (L + R) channel low-range signal 33 0h is sent to a forward orthogonal transform circuit 33 If, while the (L-R) ⁵ channel low-range signal 3 3 Oi is sent to a forward orthogonal transform circuit 33 lg.

The high-range signals 3 30c, 33 0e of the L and R channels, split by the band-splitting filters 331a, 331b, are both sent to the channel synthesis circuit 331c so as to be ¹⁰ synthesized to the (L + R) channel high-range signals 33 0g by the channel synthesis circuit 331c before being sent to a forward orthogonal transform circuit 331e.

The above-mentioned forward orthogonal transform circuits 331e, 33 lf and 33 lg process the input signals with ¹⁵ forward orthogonal transform to produce signal components 3 3 0j, 330k, 3301 which are sent to associated signal component encoding circuits 33 li, 33 lj, 331k, respectively.

The signal components 330j, 3 30k, 3301 from the forward orthogonal transform circuits 331e, 3 3 1 f, 33 lg are also sent to a control circuit 331h. The control circuit 331h determines the allocated number of bits for encoding 330m for the high-range signal components 3 3 0 j for the (L + R) channel, allocated number of bits for encoding 330n for the low-range signal components 330k for the (L + R) channel and the allocated number of bits for encoding 330o for the low-range signal components 3301 for the (L-R) channel.

Thus, the above-mentioned signal component encoding circuits 33 li, 33 lj and 331k encode the high-range side signal components 3 3 0j of the (L + R) channel, low-range signal components 330k of the (L + R) channel and the low-range signal components 3301 of the (L-R) channel, based on the numbers of allocated bits 330m, 330n and 330o determined by the control circuit 331h.

The high-range signals 330c and 330e of the L and R channels, split by the band-splitting filters 331a, 331b, are also sent to the weighting circuit 331m which then sets weighting coefficients (weighting parameters) 330s for the Land R-channels by a method which will be explained subsequently.

Using the codes 330p, 330q and 330r from the signal component encoding circuits 33 li, 33 lj and 331k and the weighting coefficient 330s from the weighting decision circuit 331m, the codestring generating circuit 3311 generates and outputs a codestring 330t.

The encoding method for the (L + R) channel may differ from that for the (L-R) channel. For example, the encoding method for the (L + R) channel may use a fixed code length to make possible encoding and decoding with a smaller processing volume, or the encoding method for the (L-R) channel may use a variable length coding to reduce the processing volume for the encoding and decoding at the cost of the increased processing volume for encoding and decoding. This enables a codestring to be constructed so that reproduction by a simplified hardware structure is possible if monaural reproduction suffices, and so that stereo reproduction is also possible. As an encoding method with a higher encoding efficiency, not only the variable-length coding, but also an encoding method for separately encoding tonal components having concentrated signal energy, may be used. It is also possible to use orthogonal transform having different transform bock lengths for the (L + R) and (L-R) channels.

Fig.33 shows a processing example of a method in which the weighting decision circuit 331m determines the weighting coefficients 330s 9parameter Rl in Fig.33).

In Fig.33, a signal energy Ehl of the L-channel highrange signals 33 0c is found at step S301 and the signal energy Eh2 of the R-channel high-range signals 330e is found at step S 3 0 2.

Then, at step S303, the sum Sh of the high-range side signal energy Ehl for the L-channel and the high-range side signal energy Eh2 for the R-channel is found. At step S304, a square root Rl of the quotient of the signal energy Ehl or Eh2 by the sum Sh is found. This value of Rl is sent as the above-mentioned weighting parameter to the codestring generating circuit 3311 where it is encoded as part of the codestring.

Fig.34 shows an example of a codestring generated by the codestring generating circuit 3311 of Fig.32.

Referring to Fig.34, the codestring is made up of a header composed of synchronization signals, number of lowrange encoding units and the above-mentioned weighting parameters Rl, low-range side encoded data of the first channel (L + R channel) (quantization fineness information, normalization coefficient information and spectral coefficients data), low-range side encoded data of the second channel (LR channel) (quantization fineness information, normalization coefficient information and spectral coefficients data) and high-range side encoded data of the (L + R) channel (quantization fineness information, normalization coefficient information and spectral coefficients data).

Fig.35 shows an illustrative structure of a decoding device adapted for decoding a codestring generated by the encoding device of Fig.32.

Referring to Fig.35, the encoded data 340a of the codestring of Fig.34 are separated by a codestring separating circuit 341a into encoded data of the high-range side (L + R) channel, encoded data of the low-range side (L + R) channel and encoded data of the low-range side (L-R) channel. The encoded data 340b of the high-range side (L + R) channel are decoded by the signal component decoding circuit 341b, while encoded data 340c of decoded by the signal the low-range side (L + R) channel are component decoding circuit 341c and the encoded data 340d of the low-range side (L-R) channel are decoded by the signal component decoding circuit 341d.

ments 340e, 340f and 340g of the highinel, low-range side (L + R) channel and R) channel, decoded by the signal circuits 341b, 341c and 341d,

The signal compo range side (L + R) chan the low-range side ( component decoding inverse orthogonal tra signal 3 40j of the (Ltransform circuit 341g respectively, are sent to associated inverse orthogonal transform circuits 341e, 341f and 341g, respectively, for inverse orthogonal transform.

The low-range signal 340i of the (L + R) channel from the tnsform circuit 34 If and the low-range R) channel from the inverse orthogonal are sent to a channel conversion circuit 3 4 li for conversion tb L-channel low-range signal 340m and R-channel low-range signal 340n. These L-channel low-range signals 340m and the R-channel low-range signals 340n are sent to associated b^nd-synthesis filters 3411 and 341m, respectively.

The high-range signal 340h of the (L + R) channel from the inverse orthogonal transform circuit 341e is sent to the weighting circuits 3 4 1 j, 341k.

The weighting circuit 34 lj is fed with the weighting coefficients 340k (weighting parameter Rl) separated from the codestring by the codestring separating circuit 341a. The is fed with the weighting coefficients eter R2) calculated by the weighting weighting circuit 341k 3401 (weighting param coefficient calculating circuit 341h from the weighting coefficients 340k. The parameter Rl) and the weighting coefficients 340k (weighting weighting coefficient 3401 (weighting parameter R2) are correlated with each other in a manner similar to the relation between the weighting parameters rl and r2 of Fig.31, and Represent weighting coefficients for the high-range side L and R channels (weighting parameters). That is, in the decoding circuit of Fig.35, weighting is made for values of the respective signals so that the sample values will be proportionate to the weighting coefficients (weighting parameters Rl and R2) associated with the L and R channels, respectively.

The (L + R) channel high-range signal 340o, corresponding inge signals 340 h weighted by the , is sent to the band-synthesis circuit annel low-range signals 340m from the :uit 34 1 i. The (L + R) channel high-range to the (L + R) high-r weighting circuit 34 1 j 3411 fed with the L-ch channel conversion cir signal 340p, corresponding to the (L + R) high-range signals

340h weighted by the band-synthesis circuit signals 340n from the channel conversion circuit 34 li.

weighting circuit 341k, is sent to the

341m fed with the R-channel low-range

4

The band-synthesis filter 3411 synthesizes the (L + R) high-range signals 340o, weighted for the L-channel, and the L-channel low-range signals 340m, by way of synthesizing the low and high ranges, to output a L-channel time-domain signal 340q of the entire range obtained by the synthesis. The bandsynthesis filter 341m synthesizes the (L + R) high-range signals 340p, weighted for the R-channel, and the R-channel lowrange signals 340n, by way of synthesizing the low and high ranges, to output a R-channel time-domain signal 340r of the entire range obtained by the synthesis.

Fig.36 shows a processing example of calculating the weighting coefficients 3401 (weighting parameters R2) by the weighting coefficient calculating circuit 341h of Fig.35.

In Fig.36, at step S401, squared value of Rl is subtracted from 1 and a square root of the resulting difference is found as R2. Meanwhile, Rl is the weighting parameter corresponding to the weighting coefficient 340k. The value of R2 is sent as the above-mentioned weighting coefficient 340p to the above-mentioned band-synthesis filter 341m.

With the illustrative method, explained with reference to Fig. Figs.32 and 35, the high-range side signal distribution can be determined by the high-range side energy proportion itself, by employing the weighting parameters Rl and R2 obtained as described above and by adding a smaller number of bits, thus enabling reproduction of the faithful stereo sound feeling by the original acoustic signals.

Noteworthy is the fact that, with the forward orthogonal transform circuit of the encoding device and the inverse orthogonal transform of the decoding circuit, the L and R channels can be used in common for the high-range sides, thus realizing saving in processing volume and buffer memory space as compared to the above-described method.

The above description has been made of a case of using a splitting filter of splitting the frequency spectrum in high and low frequency ranges. The present invention can, however, be applied to a case wherein the number of splitting of the frequency spectrum is larger than two.

Figs.37 and 38 show an illustrative structure of an encoding device (Fig.37) and a decoding device (Fig.38) when the method of the present invention is applied to the case wherein the number of splitting of the frequency spectrum is four. As explained with reference to Fig.8, the encoding unit processed with normalization is set independently of the band splitting width by the band-splitting filter. It is noted that, in distinction from the above-described method of varying the levels of the L and R channels from one encoding unit to another, the forward orthogonal transform for the high range side and the inverse orthogonal transform for the high range side can be processed as being common to the two channels, thus again realizing saving in the processing volume and the | buffer memory space.

In the encoding device shown in Fig.37, the L-channel signal 350a and the R-channel signal 350b are sent to the ⁵ band-splitting filters 351a, 351b, respectively. The bandsplitting filter 351a splits the L-channel signal 350a into four band signals 350cl, 350c2, 350c3 and 350d. The bandsplitting filter 351b similarly splits the R-channel signal 350b into four band signals 350el, 350e2, 350e3 and 350f.

¹⁰ The L-channel lowermost signal 350d, as split by the band-splitting filter 351a, and the R-channel lowermost signal 350f, as split by the band-splitting filter 351b, are both sent to the channel conversion circuit 351c.

The channel conversion circuit 351c converts the L15 channel lowermost signal 350d and the R-channel lowermost signal 350f into the (L + R) channel lowermost signal 350h and the (L-R) channel lowermost signal 3 5 Oi.

The (L + R) channel lowermost signal 350h and the (L-R) channel lowermost signal 350i are sent to a forward orthogonal transform circuits 351d, 351e, respectively. The signal components 350j, 350k from these forward orthogonal transform circuits 351d, 351e are sent to associated signal component encoding circuits 351k, 3511, respectively.

Three remaining high-range side signals 350cl, 3 50c2,

350c3 and 350el, 350e2, 350e3 of the L and R channels, as split by the band-splitting filters 351a and 351b, are sent to channel synthesis circuits 35 If 1, 35 1 f2 and 3 5 1 f3 provided in association with respective split bands. The channel synthesis circuit 35 lfl synthesizes the uppermost range side signals 350cl, 350el of the L and R channels to produce the (L + R) channel uppermost signal 350gl. The channel synthesis circuit 351f2 synthesizes the second upper range side signals 350c2, 350e2 of the L and R channels to produce the (L + R) channel second upper signal 350g2. The channel synthesis circuit 351f3 synthesizes the third upper range side signals 350c3, 350e3 of the L and R channels to produce the (L + R) channel third upper signal 350g3.

The high-range signals 350gl, 350g2 and 350g3 from the channel synthesis circuits 35 lfl, 350f2 and 350f3 are sent to forward orthogonal transform circuits 35 lgl, 351g2 and 351g3 for forward orthogonal transform. The signal components 350ml, 350m2 and 350m3, obtained on forward orthogonal transform by the forward orthogonal transform circuits 351gl, 351g2 and 351g3, respectively, are sent to the signal component encoding circuit 35 lj.

The signal components 350j, 350k, 350ml, 350m2 and 350m3 from the forward orthogonal transform circuits 35 ld, 351e, 35 lgl, 351g2 and 351g3 are sent to a control circuit 35 li. The control circuit 35 li determines the allocated numbers of bits for encoding 3 50n for the high-range side &gt;μ.

signal components 350ml, 350m2 and 350m3 of the (L + R) channel, the allocated numbers of bits for encoding 350o for the (L + R) channel lowermost signal components 350j and the allocated numbers of bits for encoding 350p for the (L-R) channel lowermost signal components 350k.

Thus, the signal component encoding circuits 35 1 j, 351k and 3511 encode the high-range side signal components of the (L + R) channel 350ml, 350m2 and 350m3 of the (L + R) channel, lower most signal components 3 5 0j of the (L + R) channel and the lower most signal components 350k of the (L-R) channel, based on the numbers of allocated bits 350n, 350o and 3 50p as determined by the control circuit 35 li.

The high-range side three-band signals 350cl, 350c2, 350c3, 350el, 350e2 and 3 50e3, obtained on splitting by the band-splitting filters 351a, 351b into four, are also sent to a weighting decision circuit 351h, which then determines the weighting coefficients (weighting parameters) 3 501 for the L and R channels by the method as described above.

The codestring generating circuit 3511 generates and output a codestring 3 50t, using codes 350q, 350r and 3 50s from the signal component encoding circuits 35 lj, 351k and 3511 and the weighting coefficients 3 501 from the weighting decision circuit 35 lh.

The decoding device shown in Fig.38 splits the encoded data 360a of the codestring generated by the encoding device of Fig.37 into encoded data of three upper side bands of the (L + R) channel, (L + R) channel of the lower most range side and the (L-R) channel lower most range side, by the codestring splitting circuit 361a. The encoded data of three upper side bands of the (L + R) channel 360hl, 360h2 and 360h3 are decoded by associated signal component decoding circuits 361gl, 361g2 and 36 lg3. The encoded data 360b of the (L + R) channel of the lower most range side are decoded by the signal component decoding circuit 361b, while the encoded data 360c of the (L-R) channel of the lower most range side are decoded by the signal component decoding circuit 3 61c.

The signal components 360il, 360Ϊ2, 360i3, 360d and 360e of the high-range side three band (L + R) channel, the lower most side (L + R) channel and the lower most side (L-R) channel, are sent to associated inverse orthogonal transform circuits 361hl, 361h2, 361h3, 361d and 361e for inverse orthogonal transform.

The lower most range (L + R) channel signal 360f from the inverse orthogonal transform circuit 361d and the lower most range (L-R) channel signal 360g from the inverse orthogonal transform circuit 361e are sent to the channel conversion circuit 36 lf for conversion into the lower most L-channel signal 360m and lower most R-channel signal 360n. The lower most L-channel signal 360m and lower most R-channel signal

360n are sent to associated weighting circuits 3611, 361m, respectively.

On the other hand, the high-range side three (L + R) channel signals 3 6 0j 1, 3 6 0j 2 and 360j3 from the inverse orthogonal transform circuits 361hl, 361h2 and 361h3 are sent to associated weighting circuits 3 6 1 j 1, 361j2, 361j3, 361kl, 361k2 and 3 6 1 k3 .

The weighting circuits 36 lj 1, 36 lj2, 36 lj3 are fed with weighting coefficients separated from the codestring 360k by the codestring dividing circuit 361a. The weighting circuits 36 lkl, 361k2 and 361k3 are fed with the weighting coefficients 360 1 calculated from the weighting coefficients 360k by the weighting coefficient calculating circuit 361i. The relation between the weighting coefficients 360k and the weighting coefficients 360 1 is similar to that between the weighting coefficients 340k, 3401 shown in Fig.35.

The high-range side three-band (L + R) channel signals 360ol, 360o2 and 360o3, obtained on weighting the highrange side three-band (L + R) channel signals 3 60j 1, 360j2 and 36 0j3 by associated weighting circuits 3 61j 1, 361j2 and

361j3, are sent to the band synthesis circuit 3611, fed with the lower most signal of the L-channel 360m from the channel conversion circuit 361f. On the other hand, the high-range side three-band (L + R) channel signals 360pl, 360p2 and

60p3, weighted by the weighting circuits 361kl, 36lk2 and

361k3, are sent to the band synthesis circuit 361m, fed with the lower most signal of the R-channel 360n from the channel conversion circuit 3 6 1 f.

The band synthesis filter 3611 synthesizes the high-range side three-band (L+R) channel signal 360ol, 360o2 and 360o3, weighted for the L-channel, and the lower most L-channel signal 360m, by way of synthesizing the low-range and highrange signals, in order to output a L-channel time-domain signal 360q for the entire range resulting from the synthesis. On the other hand, the band synthesis filter 361m synthesizes the high-range side three-band (L+R) channel signal 360pl, 360p2 and 360p3, weighted for the R-channel, and the lower most R-channel signal 360n, by way of synthesizing the lowrange and high-range signals, in order to output a R-channel time-domain signal 360r for the entire range resulting from the synthesis.

Although the foregoing description is made of the use of two channels of L and R, the present invention may be applied to acoustic signals of three or more channels. The codestring generated by the above embodiment of the encoding device, that is the encoded bitstream, can be recorded on a recording medium, such as an optical disc, a magnetic disc, a magnetic tape or a semiconductor memory, or transmitted by a transmission line, such as an optical fiber, electrical waves or infrared rays.

Although the foregoing description is made of the use of orthogonal transform, the method of the present invention may be applied to the case of using only a band-splitting filter.