Coding device and method, decoding device and method, and recording medium

Information

  • Patent Grant
  • 6614365
  • Patent Number
    6,614,365
  • Date Filed
    Wednesday, December 12, 2001
    22 years ago
  • Date Issued
    Tuesday, September 2, 2003
    20 years ago
Abstract
Coding is made possible with higher efficiency while the listener is prevented from feeling a sense of incongruity. An adaptive mixing section performs a mixing process on input signals on the basis of distortion factor information supplied from a distortion factor detection section, and controls the operation time of MS stereo coding or IS stereo coding. Furthermore, the adaptive mixing section creates power correction information in accordance with a mixing coefficient, and causes power correction to be performed during reproduction. A coding control section selects a coding method of a coding process performed in a coding section and supplies it to the coding section. The coding section selects dual coding, MS stereo coding, or IS stereo coding in accordance with the instructions from the coding control section, and codes a spectrum signal supplied from a domain conversion section.
Description




RELATED APPLICATION DATA




The present application claims priority to Japanese Application(s) No(s). P2000-380642 filed Dec. 14, 2000, which application(s) is/are incorporated herein by reference to the extent permitted by law.




BACKGROUND OF THE INVENTION




1. Field of the Invention




The present invention relates to a coding device and method, a decoding device and method, and a recording medium therefor. More particularly, the present invention relates to a coding device and method and a decoding device and method, which are capable of coding or decoding an audio signal at a low bit rate, and a recording medium therefor.




2. Description of the Related Art




In recent years, a so-called “perception audio coder (decoder)” has been developed. In a conventional CD-ROM (Compact Disk-Read Only Memory), transmission and storage of high-quality audio signals are possible at a bit rate which is approximately one twelfth the bit rate in common use.




Such a coder codes an audio signal by using a waveform portion, which is contained in the audio signal, which cannot be listened to due to the limitation of the auditory system of human beings. With regard to a stereo audio signal, for example, a coder using MS stereo coding (intermediate-portion/side-portion stereo coding) and a coder using IS stereo coding (intensity stereo coding) are known.





FIG. 1

is a block diagram showing an example of the construction of a conventional audio signal transmission system using MS stereo coding.




A left signal L and a right signal R which form a stereo audio signal is input to a computation section


1


. These signals are added by an adder


1


-


1


, and the resulting signal is output to a multiplier


1


-


2


. Meanwhile, a difference signal of those signals is generated in a subtracter


1


-


3


, and the resulting signal is output to a multiplier


1


-


4


. In the multipliers


1


-


2


and


1


-


4


, the outputs of the adder


1


-


1


and the subtracter


1


-


3


are multiplied by a coefficient x, and a sum signal M and a difference signal S are generated. These signals are coded by a coding section


2


, and are output to recording media or a transmission line


3


formed of a network, etc.




A decoding section


4


performs a decoding process on an input code sequence in order to generate a sum signal M′ and a difference signal S′. The sum signal M′ and the difference signal S′ are added by an adder


5


-


1


, and are multiplied by a coefficient y in a multiplier


5


-


2


, and the resulting signal is output as a left signal L′. Also, the sum signal M′ and the difference signal S′ are subtracted by a subtracter


5


-


3


, and the resulting signal is multiplied by a coefficient y in a multiplier


5


-


4


and is output as a right signal R′. For example, the coefficient x is set to 0.5, and the coefficient y is set to 1.0.




A sum signal exerts more influence on the sense of hearing of a human being than a difference signal. In the manner described above, by generating a sum signal M and a difference signal S and by assigning a larger amount of data (the number of bits) to the sum signal M, coding can be performed with higher efficiency than when the signals are coded (dual decoding) individually. MS stereo coding is effective for signals of lower frequency bands.





FIG. 2

is a block diagram showing an example of the construction of a conventional audio signal transmission system using IS stereo coding.




The left signal L and the right signal R which are input to a computation section


11


, are added by an adder


11


-


1


, and an intensity signal I determined by a correlation of those signals is generated. Also, a left power signal P


1


(a scaling signal in which the energy content is described) indicating the power of the left signal L and a right power signal Pr (a scaling signal in which the contents of energy are described) indicating the power of the right signal R are generated in the computation section


11


. The intensity signal I, the left power signal Pl, and the right power signal Pr are input to a coding section


12


, where the signals are coded, and thereafter, the signals are output to a transmission line


13


.




A decoding section


14


decodes the input signals, and outputs the obtained intensity signal I′, left power signal Pl′, and right power signal Pr′ to a computation section


15


. In the computation section


15


, a multiplier


15


-


1


regenerates a left signal L′ in accordance with the intensity signal I′ and the left power signal Pl′ and outputs them externally, and a multiplier


15


-


2


regenerates a right signal R′ in accordance with the intensity signal I′ and the right power signal Pr′ and outputs them externally.




As a result of performing coding by using IS stereo coding, the characteristics such that the position detection performance based on the time difference of the hearing of a human being is lower for a signal in higher-frequency domains can be used. For example, coding can be performed at a data rate approximately one half that in a case where left and right signals are coded independently.




For MS stereo coding and IS stereo coding, equivalent advantages are not obtained with respect to all the input signals. For example, MS stereo coding is an effective means only for the case where the energy of the difference signal S becomes smaller than the energy of the sum signal M. Otherwise, when the left signal L′ and the right signal R′ are regenerated from the sum signal M′ and the difference signal S′, quantization noise which occurs due to coding or decoding (quantization/inverse quantization) causes interference, and noise which can be heard clearly in the sense of hearing may be produced.




Furthermore, in IS coding, when the high-frequency components of a stereo signal are synthesized, and there is not a high correlation between a spectrum SPm which is obtained by converting the components from the time domain to the frequency domain and the envelope shapes of the original power spectra Pl and Pr, for example, when the left signal L is a signal of a trumpet and the right signal R is a signal of cymbals, the positional relationship between the respective sound sources (musical instruments) cannot be maintained, and noise which can be heard clearly may occur in the sense of hearing.




Therefore, a coding device has been conceived in which, as shown in

FIGS. 3

,


4


, and


5


, dual coding in which left and right signals are each coded independently, and MS or IS stereo coding are combined, and a coding method is selected as appropriate in accordance with an input signal.





FIG. 3

is a block diagram showing an example of the construction of a prior coding device for coding an input signal in the time domain.




A filter bank


31


-


1


divides an input left signal L(t) into signals L


n


(t), L


n−1


(t), . . . , L


1


(t) (n is the number of divided bands) of predetermined frequency bands, and outputs each signal to a corresponding dual coding section


32


and a corresponding MS/IS coding section


33


. In

FIG. 3

, although only the dual coding section


32


and the MS/IS coding section


33


for processing the signal L


n


(t) are shown, coding sections corresponding to signals L


n−1


(t), L


n−2


(t), . . . , L


1


(t) are provided in a similar manner.




Similarly to the filter bank


31


-


1


, a filter bank


31


-


2


also divides a right signal R


n


(t) into signals R


n


(t), R


n−1


(t), . . . , R


1


(t) of predetermined frequency bands, and outputs each signal to the corresponding dual coding section


32


and the corresponding MS/IS coding section


33


. In the following, when the filter bank


31


-


1


and the filter bank


31


-


2


need not be identified individually, these are referred to collectively as a filter bank


31


. The same applies to the other devices.




The dual coding section


32


codes an input signal by a dual coding method (the left signal L


n


(t) and the right signal R


n


(t) are each coded independently), and outputs the obtained data to a switch


35


. Furthermore, the dual coding section


32


creates number-of-necessary-bits information B


n


(t)


1


which is information about the amount of coded data and distortion factor information E


n


(t)


1


which is information about the distortion factor with respect to a sine wave when coding is performed, and supplies them to a coding control section


34


.




The MS/IS coding section


33


codes the input signal by the MS stereo coding method or the IS stereo coding method, and outputs the obtained data to the switch


35


. Also, the MS/IS coding section


33


creates number-of-necessary-bits information B


n


(t)


2


and distortion factor information E


n


(t)


2


, and supplies them to the coding control section


34


.




The coding control section


34


switches the contact of the switch


35


so that a code sequence which is coded by a coding method with a small distortion factor or a coding method with a smaller number of necessary bits is selected on the basis of the information supplied from the dual coding section


32


and the MS/IS coding section


33


. The code sequence selected by the switch


35


is input to a multiplexer


36


.




The multiplexer


36


combines the code sequences C


n


, C


n−1


, . . . , C


1


of each band, divided by the filter bank


31


, and outputs the combined code sequence C to a device, such as a transmission line (not shown), external of a coding device


21


.





FIG. 4

is a block diagram showing an example of the construction of a prior coding device for coding an input signal.




A domain conversion section


51


-


1


spectrum-converts the input left signal L(t) into the frequency domain, and outputs the generated spectrum signal L


n


(f) to a dual coding section


52


and an MS/IS coding section


53


. Similarly to the domain conversion section


51


-


1


, a domain conversion section


51


-


2


also spectrum-converts the input right signal R(t) into the frequency domain, and outputs the generated spectrum signal R


n


(f) to the dual coding section


52


and the MS/IS coding section


53


.




The dual coding section


52


codes the input signal by the dual coding method, and outputs the obtained code sequence to a switch


55


. Furthermore, the dual coding section


52


creates number-of-necessary-bits information B


n


(f)


1


which is information about the amount of coded data and distortion factor information E


n


(f)


1


which is information about the distortion factor with respect to a sine wave when coding is performed, and supplies them to a coding control section


54


.




The MS/IS coding section


53


codes the input signal by an MS stereo coding method or an IS stereo coding method, and outputs the obtained data to the switch


55


. Furthermore, the MS/IS coding section


53


creates number-of-necessary-bits information B


n


(f)


2


and distortion factor information E


n


(f)


2


, and supplies them to the coding control section


54


.




The coding control section


54


controls the switch


55


so that a code sequence which is coded by a coding method with a smaller distortion factor or a coding method with a smaller number of necessary bits is selected on the basis of the information supplied from the dual coding section


52


and the MS/IS coding section


53


.





FIG. 5

is a block diagram showing an example of the construction of a prior coding device in which the coding device


21


of FIG.


3


and the coding device


41


of

FIG. 4

are combined.




More specifically, in this example, the left signal L(t) and the right signal R(t) are divided into a predetermined number of bands by filter banks


71


-


2


and


71


-


2


, and the divided signals are spectrum-converted by domain conversion sections


72


-


1


and


72


-


2


, respectively. The converted spectrum signals are coded by a dual coding section


73


and an MS/IS coding section


74


. In a coding control section


75


and a switch


76


, among the code sequences coded in the dual coding section


73


and the MS/IS coding section


74


, the code sequence by the coding method with higher efficiency (with a smaller distortion factor or with a smaller amount of data) is selected and is output to a multiplexer


77


. Then, after the input data of all the bands is combined by the multiplexer


77


, the data is output to outside a coding device


61


.




Next, referring to the flowchart in

FIG. 6

, the process of the coding control section


34


of the coding device


21


of

FIG. 3

will be described below. Although descriptions are omitted, the processes of the coding control section


54


of FIG.


4


and the coding control section


75


of

FIG. 5

are the same as the above. In this example, it is assumed that the coding control section


34


selects a coding method on the basis of the distortion factor.




In step S


1


, the coding control section


34


compares the distortion factor information E


n


(t)


1


supplied from the dual coding section


32


with the distortion factor information E


n


(t)


2


supplied from the MS/IS coding section


33


. Then, the coding control section


34


determines whether or not the distortion factor supplied from the dual coding section


32


is smaller than the distortion factor supplied from the MS/IS coding section


33


. When it is determined that the distortion factor is smaller, in step S


3


, the coding control section


34


controls the switch


35


so that the data coded by the dual coding section


32


is output to the multiplexer


36


.




When, on the other hand, it is determined in step S


2


that the distortion factor supplied from the dual coding section


32


is greater than the distortion factor supplied from the MS/IS coding section


33


, the process proceeds to step S


4


, where the coding control section


34


controls the switch


35


so that the data coded by the MS/IS coding section


33


is output to the multiplexer


36


.




The same process is performed in the other bands. As a result, a code sequence C which is coded for each band by a low-bit-rate coding method is created, and is output to outside the coding device


21


.




In the manner described above, the coding efficiencies of the respective coding methods are compared with each other, and an optimum method is selected according to the result thereof, thereby making it possible to obtain coded data at a lower bit rate in comparison with a case in which coding is performed by a single coding method.





FIGS. 7A

,


7


B,


7


C, and


7


D show an example of the relationship among the operation time probability P


MS


of MS stereo coding or the operation time probability P


IS


of IS stereo coding in the coding devices of

FIGS. 3

to


5


, the signal power to noise power ratio SNR of the coded (quantized) signal, and the separation of the left and right signals.




As shown in

FIG. 7A

, the probability P


MS


or P


IS


shown in the horizontal axis is proportional to the SNR shown in the vertical axis. The nearer the probability P


MS


or P


IS


approaches 100% (monaural), the more the SNR is improved.





FIG. 7B

shows the change in the probability P


MS


or P


IS


with respect to time.

FIG. 7C

shows the change in the SNR with respect to time. As shown in these figures, since the waveforms thereof become in same phase, and the coding efficiency is improved by increasing the probability P


MS


or P


IS


in accordance with the input signal, the SNR is also improved, and thus the sound quality is improved. For this reason, it is preferable from the viewpoint of coding efficiency that the probability P


MS


or P


IS


be higher.




However, high probability P


MS


indicates that there is a high correlation between the left and right signals. High probability P


IS


indicates that the intensity signal and the spectrum to be coded are for one channel although the power levels are different. That is, high probability P


MS


or P


IS


is indicates that a stereo signal is changed into a monaural signal. As shown in

FIG. 7D

, the separation of the left and right signals becomes poorer as the probability P


MS


/P


IS


is increased.




Furthermore, since the probability P


MS


or P


IS


is linked with the SNR, if the value of the probability P


MS


or P


IS


is high, there is the risk that, due to a change of the properties of the input signal or due to a change of the input signal with respect to time, the SNR falls below the perceptible noise level limit in an auditory psychological model (a level at which, if the SNR decreases to less than that level, perceptual noise is heard). Therefore, when considered together, the value of the probability P


MS


or P


IS


being high is not always preferable.




In the coding devices shown in

FIGS. 3

to


5


, a determination of whether the efficiency when coding is performed by MS stereo coding or IS stereo coding or the efficiency when coding is performed by dual coding is superior, cannot be known until the two coding processes are actually performed, thus presenting the problem that the amount of processing in each coding section increases.




Also, when MS stereo coding or IS stereo coding is performed, the coding efficiency can be increased (quantized noise can be decreased). However, when it is not performed, such advantages cannot be obtained. Consequently, sound-quality variations with respect to time are large between when MS stereo coding or IS stereo coding is performed or not, and a problem arises in that the listener feels a substantial sense of incongruity in the sense of hearing.




SUMMARY OF THE INVENTION




The present invention is made in view of such circumstances. The present invention aims to code or decode an audio signal at a higher efficiency while the listener is prevented from feeling a sense of incongruity.




To this end, according to one aspect of the present invention, there is provided a coding device for coding an input signal, comprising: coding method selection means for selecting a coding method in accordance with the input signal; coding means for coding the input signal in accordance with the coding method selected by the coding method selection means; distortion factor detection means for detecting a distortion factor of coding by the coding means; and mixing means for mixing the left and right components of the input signal on the basis of a mixing ratio determined in such a manner as to correspond to the distortion factor detected by the distortion factor detection means, wherein the coding method selection means selects the coding method in accordance with the input signal mixed by the mixing means.




The coding device may further comprise output correction information creation means for creating output correction information which is used when the input signal coded by the coding means is decoded.




The coding method selection means may select the coding method for the input signal on the basis of a threshold value determined according to the construction of the coding device.




The coding method selection means may select the coding method from among a dual coding method, an MS stereo coding method, and an IS stereo coding method.




The coding method selection means may select the dual coding method to perform coding on the basis of the correlation between the left and right components of the input signal, that is, the total of the sum signals with respect to the total of the difference signals of the left and right components, and may select MS stereo coding or IS stereo coding to perform coding on the basis of the maximum value of the absolute value of the difference of the left and right components of the input signal.




The mixing means may store the mixing ratio, and may change the mixing ratio on the basis of an interpolation function of the mixing ratio determined immediately before and the mixing ratio determined currently.




The coding device may further comprise input signal storage means for storing the input signal, wherein the mixing means may mix again the left and right components of the same input signal on the basis of the distortion factor used when the input signal is coded.




According to another aspect of the present invention, there is provided a coding method for coding an input signal, comprising: a coding method selection step of selecting a coding method in accordance with the input signal; a coding step of coding the input signal in accordance with the coding method selected in the coding method selection step; a distortion factor detection step of detecting a distortion factor of coding in the coding step; and a mixing step of mixing the left and right components of the input signal on the basis of a mixing ratio determined in such a manner as to correspond to the distortion factor detected in the distortion factor detection step, wherein the process of the coding method selection step selects the coding method in accordance with the input signal mixed in the mixing step.




According to another aspect of the present invention, there is provided a recording medium having recorded thereon a computer-readable program, the program comprising: a coding method selection step of selecting a coding method in accordance with an input signal; a coding step of coding the input signal in accordance with the coding method selected in the coding method selection step; a distortion factor detection step of detecting a distortion factor of coding in the coding step; and a mixing step of mixing the left and right components of the input signal on the basis of a mixing ratio determined in such a manner as to correspond to the distortion factor detected in the distortion factor detection step, wherein the process of the coding method selection step selects the coding method in accordance with the input signal mixed in the mixing step.




According to another aspect of the present invention, there is provided a decoding device for decoding a code sequence coded by a predetermined coding method, the decoding device comprising: decoding method selection means for selecting a decoding method corresponding to the coding method; decoding means for decoding an input code sequence in accordance with the decoding method selected by the decoding method selection means; correction means for correcting the left and right components of a signal decoded by the decoding means on the basis of information supplied from the coding device; and output means for outputting the signal corrected by the correction means.




According to another aspect of the present invention, there is provided a decoding method for decoding a code sequence coded by a predetermined coding method, the decoding method comprising: a decoding method selection step of selecting a decoding method corresponding to a coding method used by a coding device; a decoding step of decoding an input code sequence in accordance with the decoding method selected in the decoding method selection step; a correction step of correcting the left and right components of a signal decoded in the decoding step on the basis of information supplied from the coding device; and an output step of outputting the signal corrected in the correction step.




According to another aspect of the present invention, there is provided a recording medium having recorded thereon a computer-readable program, the program comprising: a decoding method selection step of selecting a decoding method corresponding to a coding method used by a coding device; a decoding step of decoding an input code sequence in accordance with the decoding method selected in the decoding method selection step; a correction step of correcting the left and right components of a signal decoded in the decoding step on the basis of information supplied from the coding device; and an output step of outputting the signal corrected in the correction step.




In the coding device and method and the program of the recording medium of the present invention, a coding method is selected in accordance with an input signal, the input signal is coded on the basis of the selected coding method, and the left and right components of the input signals are mixed. Furthermore, a coding method is selected in accordance with the mixed input signals. Therefore, it is possible to code an audio signal with higher efficiency.




In the decoding device and method and the program of the recording medium of the present invention, a decoding method corresponding to a coding method used by a coding device is selected, and an input code sequence is decoded on the basis of the selected decoding method. Furthermore, the left and right components of the decoded signal are corrected on the basis of the information supplied from the coding device, and the corrected signal is output. Therefore, it is possible to reproduce a coded audio signal with higher efficiency while the listener is prevented from feeling a sense of incongruity.




Further objects, features and advantages of the present invention will become apparent from the following description of the preferred embodiments with reference to the attached drawings.











BRIEF DESCRIPTION OF THE DRAWINGS





FIG. 1

is a block diagram showing an example of the configuration of a prior audio signal transmission system employing MS stereo coding;





FIG. 2

is a block diagram showing an example of the configuration of a prior audio signal transmission system employing IS stereo coding;





FIG. 3

is a block diagram showing an example of the construction of a prior coding device;





FIG. 4

is a block diagram showing an example of the construction of another prior coding device;





FIG. 5

is a block diagram showing an example of the construction of another prior coding device;





FIG. 6

is a flowchart illustrating the process of a prior coding device;





FIGS. 7A

,


7


B,


7


C, and


7


D show the relationship between the operation of the prior coding device and a signal to be generated;





FIG. 8

is a block diagram showing an example of the construction of a coding device to which the present invention is applied;





FIG. 9

is a block diagram showing an example of the construction of an adaptive mixing section of

FIG. 8

;





FIG. 10

is a table showing an example of information stored in a mixing coefficient setting section of

FIG. 9

;





FIG. 11

is a table showing an example of information stored in a power correction section of

FIG. 9

;





FIG. 12

shows an example of the construction of a multiplier of

FIG. 9

;





FIG. 13

shows an example of an interpolation function of a mixing coefficient;





FIG. 14

is a block diagram showing an example of the construction of a coding control device of

FIG. 8

;





FIG. 15

is a flowchart illustrating the process of the coding device of

FIG. 8

;





FIG. 16

is a flowchart illustrating the details of a process performed in step S


12


of

FIG. 15

;





FIG. 17

is a flowchart illustrating the details of a process performed in step S


14


of

FIG. 15

;





FIGS. 18A

,


18


B,


18


C, and


18


D show the relationship between the operation of the coding device of

FIG. 8 and a

signal to be generated;





FIG. 19

is a block diagram showing an example of the construction of a decoding device to which the present invention is applied;





FIG. 20

is a block diagram showing an example of the construction of a power weighting section of

FIG. 19

;





FIG. 21

is a block diagram showing an example of the construction of a multiplier of

FIG. 20

;





FIG. 22

shows an example of an interpolation function of a power weighting coefficient;





FIG. 23

is a flowchart illustrating the process of the decoding device of

FIG. 19

;





FIG. 24

is a flowchart illustrating the details of a process performed in step S


74


of

FIG. 23

; and





FIG. 25

is a block diagram showing an example of the configuration of a personal computer.











DESCRIPTION OF THE PREFERRED EMBODIMENTS





FIG. 8

is a block diagram showing an example of the construction of a coding device to which the present invention is applied.




A filter bank


101


-


1


divides a left signal L(t) within an input audio signal into signals L


n


(t), L


n−1


(t), . . . , L


1


(t) of n frequency bands, and outputs the generated signal L


n


(t) to an adaptive mixing section


102


. Also, similarly to the filter bank


101


-


1


, a filter bank


101


-


2


divides a right signal R(t) within the input audio signal into signals R


n


(t), R


n−1


(t), . . . , R


1


(t) of n frequency bands, and outputs the generated signal R


n


(t) to the adaptive mixing section


102


. Although not shown, for the signals L


n−1


(t), . . . , L


1


(t) and R


n−1


(t), . . . , R


1


(t), corresponding processing sections are also provided.




The adaptive mixing section


102


performs a mixing process on the signals L


n


(t) and R


n


(t) on the basis of distortion factor information E


n


(f) supplied from a distortion factor detection section


106


in order to generate signals L


n


(t)


mix


and R


n


(t)


mix


(the details thereof will be described later with reference to FIG.


9


). The generated signals L


n


(t)


mix


and R


n


(t)


mix


are supplied to domain conversion sections


103


-


1


and


103


-


2


, respectively. As will be described later, since the distortion factor detection section


106


generates distortion factor information E


n


(f) according to the results of the coding in a coding section


105


, the mixing ratio is set to 0 in the initial state of the operation. That is, a mixing process is not performed on the signals L


0


(t) and R


0


(t).




Furthermore, the adaptive mixing section


102


creates power correction information P


n,adj


(t) for correcting the output of the left and right signals, and outputs it to a multiplexer


107


.




The domain conversion section


103


-


1


performs domain conversion, such as MDCT (Modified Discrete Cosine Transform), on the supplied signal L


n


(t)


mix


, and outputs the generated spectrum signal L


n


(f) to a coding control section


104


and the coding section


105


. Similarly, a domain conversion section


103


-


2


performs domain conversion on the supplied signal R


n


(t)


mix


and outputs the generated spectrum signal R


n


(f) to the coding control section


104


and the coding section


105


.




The coding control section


104


selects a coding method for the coding process performed by the coding section


105


on the basis of the spectrum signals L


n


(t) and R


n


(f) supplied from the domain conversion section


103


, so that the coding section


105


is controlled.




The coding section


105


selects dual coding, MS stereo coding, or IS stereo coding under the control of the coding control section


104


, codes the spectrum signals L


n


(t) and R


n


(f) supplied from the domain conversion section


103


, and outputs the obtained data sequence C


n


to the multiplexer


107


. The above processing is performed on the signals L


n−1


(t), . . . , L


1


(t) and R


n−1


(t), . . . , R


1


(t) in a similar manner.




The multiplexer


107


combines a code sequence C


n


of a predetermined band, supplied from the coding section


105


with the code sequences C


n−1


, . . . , C


1


of the other bands, and outputs the combined audio data C to a device (not shown) provided external to a coding device


91


, a network, etc. The combined audio data C contains power correction information P


n,adj


(t) supplied from the adaptive mixing section


102


and information indicating by which coding method the signals are coded.





FIG. 9

is a block diagram showing a detailed example of the construction of the adaptive mixing section of FIG.


8


.




A power computing section


121


computes the power values Pl


n


and Pr


n


from the signals L


n


(t) and R


n


(t) which are divided into predetermined bands by the filter banks


101


-


1


and


101


-


2


, respectively, and outputs them to a power correction section


123


.




A mixing coefficient setting section


122


extracts mixing coefficients from a table stored in a built-in storage section corresponding to the distortion factor information E


n


(f) supplied from the distortion factor detection section


106


, and sets a mixing coefficient a of multipliers


124


-


1


and


124


-


2


and a mixing coefficient b of multipliers


125


-


1


and


125


-


2


. Furthermore, the mixing coefficient setting section


122


supplies the extracted mixing coefficients a and b to the power correction section


123


.




The multipliers


124


-


1


and


124


-


2


multiply the input signals L


n


(t) and R


n


(t) by the mixing coefficient a which is set by the mixing coefficient setting section


122


, and outputs the obtained signal to adders


126


-


1


and


126


-


2


, respectively. The multipliers


125


-


1


and


125


-


2


multiply the input signals R


n


(t) and L


n


(t) by the mixing coefficient b which is set by the mixing coefficient setting section


122


, and outputs the obtained signal to the adders


126


-


1


and


126


-


2


, respectively.




The adder


126


-


1


adds together the left signal Ln(t) with which the coefficient a is multiplied by the multiplier


124


-


1


and the right signal Rn(t) with which the coefficient b is multiplied by the multiplier


125


-


1


, and outputs the added result, as a signal L


n


(t)


mix


, to the domain conversion section


103


-


1


. Also, the adder


126


-


2


adds together the right signal Rn(t) with which the coefficient a is multiplied by the multiplier


124


-


1


and the left signal Ln(t) with which the coefficient b is multiplied by the multiplier


125


-


2


, and outputs the added result, as a signal R


n


(t)


mix


, to the domain conversion section


103


-


2


.





FIG. 10

shows an example of a correspondence table of distortion factor information E


n


(f), stored in a storage section (not shown) of the mixing coefficient setting section


122


and the mixing coefficients a and b.




In this example, the distortion factor information E


n


(f) is expressed as a percentage, and hereinafter this value will be referred to as “E”. For example, E=0% means that the perceptible noise is zero. Also, E=100% means that the noise is at a perceptible level in all the spectral domains.




In this example, mixing coefficients a=1.00 and b=0.00 are set in such a manner as to correspond to the distortion factor E=0%. In this case, since the input left and right signals L


n


(t) and R


n


(t) are not mixed, coding is performed in a completely separated state (completely stereo). Also, mixing coefficients a=0.50 and b=0.50 are set in such a manner as to correspond to the distortion factor E=100%. In this case, the input left and right signals L


n


(t) and R


n


(t) are mixed at the same ratio, and coding is performed in a completely unified state (completely monaural).




The power correction section


123


creates power correction information P


n,adj


(t) which is used when power correction is performed in a decoding device (

FIG. 19

) (to be described later) on the basis of the power values Pl


n


and Pr


n


supplied from the power computing section


121


and the mixing coefficients a and b supplied from the mixing coefficient setting section


122


, and outputs them to the multiplexer


107


. That is, the power correction section


123


has stored, in a storage section (not shown), the correspondence table in which the relationships among the power correction information P


n,adj


(t), the mixing coefficients a and b, and the power values Pl


n


and P


rn


.





FIG. 11

shows an example of the correspondence table stored in the power correction section


123


.




In this example, the power values Pl


n


and Pr


n


computed in the power computing section


121


, the distortion factor information E


n


(f), the mixing coefficients a and b, the power values Pl


nmix


and Pr


nmix


of signals L


n


′(t)


mix


and R


n


′(t)


mix


to be regenerated in the decoding device


151


, and the power correction information P


n,adj


(t) are made to correspond to each other. In this example, the power correction information P


n,adj


(t) is represented using power weighting coefficients c and d which are set in the decoding device


151


.




For example, as shown in the second row of

FIG. 11

, when the power value of the signal L


n


(t) is Pl


n


=1.0, the power value of the signal R


n


(t) is Pr


n


=1.0, and the distortion factor E=0%, the mixing coefficients are set as a=1.00 and b=0.00 from the correspondence table shown in FIG.


10


. The power value of the signal L


n


′(t)


mix


in the decoding device


151


is set as Pl


nmix


=1.0 and the power value of the signal R′


n


(t)


mix


is set as Pr


nmix


=1.0. Since the power correction information P


n,adj


(t) contains a coefficient which causes the regenerated signal to approach the input signal, the coefficient for correcting the power of the signal L′


n


(t)


mix


is set to c=1.00, and the coefficient for correcting the power of the signal R′


n


(t)


mix


is set to d=1.00.





FIG. 12

is a block diagram showing a detailed example of the construction of the multiplier


124


-


1


(although not shown, the multiplier


124


-


2


is also similarly constructed).




In this example, buffers


124


A and


124


B are provided. At the current time (time t=0), the set mixing coefficient a(t


0


) is stored in the buffer


124


A, and the mixing coefficient a(t


1


) which was set immediately before (which has been set at time t=1) is stored in the buffer


124


B.




When the mixing coefficient is changed, there are cases in which a noncontinuous point occurs in the signal which is output at that time. Therefore, as indicated in curves i to iii of

FIG. 13

, the occurrence of a noncontinuous point can be prevented by changing the mixing coefficient in a manner of a straight line or in a manner of a curve. Although in this example, two buffers are provided, three or more buffers may be provided. A degree of the interpolation function which interpolates each mixing coefficient may be one, two, three, etc. Of course, similarly, a buffer may be provided in multipliers


125


-


1


and


125


-


2


, so that the mixing coefficient b is stored and the mixing coefficient is changed on the basis of the interpolation function.





FIG. 14

is a block diagram showing a detailed example of the construction of the coding control device


104


of FIG.


8


.




A normalization section


141


-


1


normalizes the spectrum signal L


n


(f) input from the domain conversion section


103


-


1


for each divided frequency band or for each range of a small domain in which spectra within the same divided frequency band are collected at several spectral signal in order to generate a normalized spectrum signal l


n


(f), and outputs it to adders


142


-


1


and


142


-


2


. Similarly, the adder


142


-


2


normalizes the spectrum signal R


n


(f) input from the domain conversion section


103


-


2


in order to generate a normalized spectrum signal r


n


(f), and outputs it to the adders


142


-


1


and


142


-


2


. The normalized spectrum signals l


n


(f) and r


n


(f) are added together or the normalized spectrum signal l


n


(f) is subtracted he normalized spectrum signal r


n


(f) in the spectrum in the adders


142


-


1


and


142


-


2


, respectively, and the generated signals s


n


(f)(=|l


n


(f)+r


n


(f)|) and d


n


(f)(=|l


n


(f)−r


n


(f)|) are supplied to a comparator


143


.




The comparator


143


computes the total sum values S and D for each divided frequency band of each of the input signals sn and dn, and selects, based on the ratio S/D thereof, the coding method for the spectrum signals L


n


(f) and R


n


(f), performed in the coding section


105


. In the comparator


143


, it is determined whether or not coding should be performed by dual coding. Which one of MS stereo coding and IS stereo coding is used to code the spectrum signals L


n


(f) and R


n


(f) is determined in a comparator


144


(to be described later).




The comparator


144


, based on the difference components d


n


(f)(=l


n


(f)−r


n


(f)) of the normalized spectrum signals l


n


(f) and r


n


(f) supplied from the comparator


143


, selects a coding method from MS stereo coding and IS stereo coding, which is to be used to code the spectrum signals L


n


(f) and R


n


(f).




Next, the operation of the coding device


91


of

FIG. 8

will be described with reference to the flowchart in FIG.


15


.




In step S


11


, a filter bank


101


divides an input audio signal for each predetermined frequency band, and outputs the generated signals to the adaptive mixing section


102


. That is, the filter bank


101


-


1


divides the left signal L(t) into n bands, and outputs the left signal L


n


(t) to the adaptive mixing section


102


. Also, the filter bank


101


-


2


divides the right signal R(t) into n bands, and outputs the right signal R(t) to the adaptive mixing section


102


.




In step S


12


, the adaptive mixing section


102


performs a mixing process on the input signals L


n


(t) and R


n


(t) on the basis of the distortion factor information E


n


(f) supplied from the distortion factor detection section


106


. The details of the mixing process will be described later with reference to the flowchart in FIG.


16


.




The signals L


n


(t)


mix


and R


n


(t)


mix


generated by the mixing process are supplied to the domain conversion section


103


. In step S


13


, these signals are converted from the time domain to the frequency domain by MDCT, etc., and the spectrum signals L


n


(t) and R


n


(f) after conversion are output to the coding control section


104


and the coding section


105


.




In step S


14


, the coding control section


104


performs a process for controlling the coding method of the spectrum signals L


n


(f) and R


n


(f) input to the coding section


105


. The details of the coding control process will be described later with reference to the flowchart in FIG.


17


.




In step S


15


, the coding section


105


selects dual coding, MS stereo coding, or IS stereo coding in accordance with the instructions from the coding control section


104


, codes the spectrum signals L


n


(f) and R


n


(f) supplied from the domain conversion section


103


in accordance with the selected method, and outputs the obtained code sequence C


n


to the multiplexer


107


. Which coding method was used to code the signals is uniquely determined in the decoding device


151


, for example, in accordance with a combination of information for identifying a codebook to which a reference is made, information about the accuracy of quantization, the normalization information, etc., when a spectrum signal is coded.




The distortion factor detection section


106


detects the distortion factor of the coding process performed in the coding section


105


, and creates distortion factor information E


n


(f). The created distortion factor information E


n


(f) is supplied to the adaptive mixing section


102


in step S


16


, and is used for processing in step S


16


and subsequent steps. The above processing is performed in all bands.




In step S


17


, the multiplexer


107


combines the code sequence C


n


supplied from the coding section


105


with the code sequences C


n−1


, C


n−2


, . . . , C


1


from the coding sections of the other bands, and outputs the obtained code sequence C to a device (not shown) provided external to the coding device


91


or outputs it to a network, etc. The code sequence C contains information, such as power correction information P


n,adj


(t) supplied from the adaptive mixing section


102


.




Next, referring to the flowchart in

FIG. 16

, a description will be given of the mixing process of the adaptive mixing section


102


performed in step S


12


of FIG.


15


.




In step S


31


, the mixing coefficient setting section


122


determines whether or not distortion factor information E


n


(f) is supplied from the distortion factor detection section


106


. When it is determined that the distortion factor information E


n


(f) is supplied, the process proceeds to step S


32


, where the mixing coefficients a and b of the multipliers


124


and


125


are set on the basis of the distortion factor information E


n


(f). When, for example, the fact that the distortion factor E is 100% is supplied, the mixing coefficient setting section


122


extracts mixing coefficients a=0.95 and b=0.05 from the correspondence table such as that shown in

FIG. 10

, sets the mixing coefficient a of the multiplier


124


to 0.95, and sets the mixing coefficient b of the multiplier


125


to 0.05. The mixing coefficient setting section


122


supplies the set mixing coefficients to the power correction section


123


.




On the other hand, when it is determined in step S


31


that the distortion factor information E


n


(f) is not supplied from the distortion factor detection section


106


, in step S


33


, the mixing coefficient setting section


122


sets the initial mixing coefficients in the multipliers


124


and


125


, respectively. That is, as described above, in the initial state, the distortion factor E is set to 100%, and the mixing coefficients a and b are set to 1.00 and b=0.00, respectively.




In step S


34


, the adder


126


-


1


adds together the signal obtained by multiplying the left signal L


n


(t) by the mixing coefficient a in the multiplier


124


-


1


and the signal obtained by multiplying the right signal R


n


(t) by the mixing coefficient b in the multiplier


125


-


1


, generates a mixing signal L


n


(t)


mix


, and outputs it to the domain conversion section


103


-


1


.




In step S


35


, the adder


126


-


2


adds together the signal obtained by multiplying the right signal R


n


(t) by the mixing coefficient a in the multiplier


124


-


2


and the signal obtained by multiplying the left signal L


n


(t) by the mixing coefficient b in the multiplier


125


-


2


, generates a mixing signal R


n


(t)


mix


, and outputs it to the domain conversion section


103


-


2


.




More specifically, when the above-described mixing coefficients (a=0.95 and b=0.05) are set in the multipliers


124


and


125


in steps S


34


and S


35


, one of the left and right signals L


n


(t) and R


n


(t) is output to the domain conversion section


103


after 5% of the other is mixed. Also, in the case of the initial state, and the signals are output to the domain conversion section


103


in a completely stereo state in which the left and right signals L


n


(t) and R


n


(t) are not mixed.




In step S


36


, the power computing section


121


computes the power values Pl


n


, and Pr


n


of the signals L


n


(t) and R


n


(t) which are divided into predetermined bands by the filter bank


101


, and supplies the power values to the power correction section


123


.




In step S


37


, the power correction section


123


creates power correction information P


n,adj


(t) which is used when power correction is performed in the decoding device


151


(to be described later) (see

FIG. 19

) on the basis of the power values Pl


n


and Pr


n


of the signals L


n


(t) and R


n


(t) supplied from the power computing section


121


and the mixing coefficients a and b supplied from the mixing coefficient setting section


122


, and outputs them to the multiplexer


107


.




For example, when the fact that the power value Pl


n


of the signal L


n


(t) is 5.0 and the power value Pr


n


of the signal R


n


(t) is 1.0 is supplied from the power computing section


121


and the fact that the mixing coefficients a=0.75 and b=0.25 is supplied from the mixing coefficient setting section


122


(in the case of the distortion factor E=50%), as indicated in the fourth row from the top in

FIG. 11

, then c=1.25 and d=0.50 are extracted as the power correction information P


n,adj


(t) (power weighting coefficient). That is, in the decoding device


151


, since the signal L′


n


(t)


mix


, which is obtained when the data of the signal L


n


(t) is decoded, is reproduced with the power value Pl


nmix


=4.0 and the signal R′


n


(t)


mix


, which is obtained when the data of the signal R


n


(t) is decoded, is reproduced with the power value Pr


nmix


=2.0, power weighting coefficients c and d, which become equal to the input signal when these are multiplied by the regenerated signal, are extracted, and these are output to the multiplexer


107


.




For example, when the distortion factor is high, the adaptive mixing section


102


sets the mixing coefficient so that the left and right signals are changed in a monaural manner, so that the operation probability of the MS stereo coding or the IS stereo coding is increased. As a result, the SNR can be increased, and the distortion factor can be decreased. Furthermore, as described above, as a result of setting the mixing coefficient on the basis of the feedback distortion factor information, a region having a high correlation is created in a region where there is not a high correlation in the regions of the normalized spectrum signals l


n


(f) and r


n


(f). Furthermore, in the decoding device, since power correction is performed based on the power correction information P


n,adj


(t), the separation of the left and right signals is maintained.




Next, referring to the flowchart in

FIG. 17

, a description will be given of the coding control process of the coding control section


104


performed in step S


14


of FIG.


15


.




In step S


51


, the normalization section


141


normalizes the input signal for each divided frequency band or for each range of a small domain in which spectra within the same divided frequency band are collected at several spectral signal. The generated normalized spectral signals l


n


(f) and r


n


(f) are supplied to the adder


142


-


1


and the subtracter


142


-


2


. In step S


52


, the sum signal s


n


(f)(=|l


n


(f)+r


n


(f)|) of the normalized spectrum signals is generated by the adder


142


-


1


, and the difference signal d


n


(f)(=|l


n


(f)−r


n


(f)|) is generated by the subtracter


142


-


2


. The generated sum signal s


n


(f) and the generated difference signal d


n


(f) of the normalized spectrum signals are supplied to the comparator


143


.




In step S


53


, the comparator


143


computes the total sum value S of all the bands of the input signal s


n


(f) on the basis of the following equation (1) and computes the total sum value D in the range where the signal d


n


(f) is normalized on the basis of the following equation (2):









S
=




f
=

f
0



f

1
-
1









&LeftBracketingBar;


S
n



(
f
)


&RightBracketingBar;






(
1
)






D
=




f
=

f
0



f

1
-
1









&LeftBracketingBar;


d
n



(
f
)


&RightBracketingBar;






(
2
)













where f0 indicates the start spectrum number in the normalized range, and f1 indicates the end spectrum number.




The more similar (the higher the correlation) the normalized spectrum signal l


n


(f) and the normalized spectrum signal r


n


(f) are to each other, the larger the total sum value S and the smaller the total sum value D. In contrast, when the normalized spectrum signal l


n


(f) and the normalized spectrum signal r


n


(f) differ from each other (the correlation is lower), since the total sum value S and the total sum value D become substantially the same values, by computing the ratio of the total sum values S and D (total sum value ratio S/D), the correlation between the normalized spectrum signal l


n


(f) and the normalized spectrum signal r


n


(f) can be obtained. For example, when the value of the total sum value ratio S/D is greater than “1”, this indicates that the correlation between the normalized spectrum signal l


n


(f) and the normalized spectrum signal r


n


(f) is high.




Then, in step S


54


, the comparator


143


determines whether or not the total sum value ratio S/D computed in step S


53


is smaller than a permissible error level (threshold value) Thr which is set in advance for each divided frequency band or for each small normalized domain. When it is determined by the comparator


143


that the total sum value ratio S/D is smaller than the permissible error level Thr, the process proceeds to step S


55


, where a selection is made such that the spectrum signals L


n


(f) and R


n


(f) input to the coding section


105


are coded by dual coding, and this is supplied to the coding section


105


. That is, the permissible error level is set so that if the total sum value ratio S/D is equal to or greater than a predetermined level (if there is a correlation over a predetermined level between the normalized spectrum signal l


n


(f) and the normalized spectrum signal r


n


(f)), coding is forcedly performed by MS or IS stereo coding. In this embodiment, the correlation between the normalized spectrum signal l


n


(f) and the normalized spectrum signal r


n


(f) is determined by using the ratio of the total sum value S to D. However, of course, the correlation determination method is not limited to this, and the determination may be performed by using another parameter, such as a correlation coefficient being obtained by comparing the absolute value of l


n


(f) with that of r


n


(f).




On the other hand, when it is determined in step S


54


that the total sum value ratio S/D is equal to or greater than the permissible error level Thr, the comparator


143


supplies that fact to the comparator


144


. Then, in step S


56


, the comparator


144


determines whether or not the maximum value of d


n


(f) with respect to the spectrum of the target band is greater than the quantization accuracy level which can be realized by the decoding device


151


. That is, the comparator


144


selects MS stereo coding when the difference signal d


n


(f) needs to be coded, and when the sum signal d


n


(f) need not to be coded, the comparator


144


selects IS stereo coding.




When it is determined in step S


56


by the comparator


144


that the maximum value of d


n


(f) is greater than the quantization accuracy level Thq, the process proceeds to step S


57


, where a selection is made such that the spectrum signals L


n


(f) and R


n


(f) input to the coding section


105


are coded by MS stereo coding, and this is supplied to the coding section


105


. Also, when it is determined in step S


56


by the comparator


144


that the maximum value of d


n


(f) is equal to or smaller than the quantization accuracy level Thq, the process proceeds to step S


58


, where a selection is made such that the spectrum signals L


n


(f) and R


n


(f) input to the coding section


105


are coded by IS stereo coding is selected, and this is supplied to the coding section


105


.




As a result, even if there is a high correlation between the normalized spectrum signal l


n


(f) and the normalized spectrum signal r


n


(f), and even if there is a possibility that a higher SNR can be realized by dual coding than MS or IS stereo coding, when the total sum value ratio S/D is higher than the threshold value at which hearing as noise is not possible, the input signal is coded by MS or IS stereo coding.




Furthermore, even when the difference signal d


n


(f) is not coded, since the information about the normalization of the left and right signals is coded, IS stereo coding can be considered as being equivalent to MS stereo coding. As a result, there is no need to separately provide a processing section for performing MS stereo coding and a processing section for performing IS stereo coding, and the coding device


91


can be formed to be smaller.




The permissible error level Thr is set according to the construction of the coding system, such as the block length of domain conversion and bit allocation. And, for the quantization accuracy level Thq, a highest quantization accuracy level which can be realized by the coding device


91


may be set, or a quantization accuracy level Thq(f) may be set for each frequency band. That is, similarly to the permissible error level Thr, the quantization accuracy level Thq is also set according to the system.





FIG. 18A

shows the relationship between the separation and the signal-to-noise ratio SNR in the coding device


91


.

FIG. 18B

shows the change in the signal-to-noise ratio SNR of the coded (normalized) signal with respect to time.

FIG. 18C

shows the change in the operation time probability P


MS


of MS stereo coding or the change in the operation time probability P


IS


of IS stereo coding with respect to time.

FIG. 18D

shows the change in the separation of the left and right signals L


n


(t) and R


n


(t) signals with respect to time.




As shown in

FIGS. 18B and 18C

, since the signal-to-noise ratio SNR is linked with the operation time probability P


MS


of MS stereo coding or the operation time probability P


IS


of IS stereo coding, by varying the mixing coefficient appropriately as described above, SNR can be improved by controlling the probability P


MS


or P


IS


. This makes it possible to improve the sound quality.




And, as shown in

FIG. 18A

, as the SNR is improved, the separation of the left and right signals becomes poorer (becomes to be monaural). Consequently, as shown in

FIG. 18D

, the separation becomes poorer in response to the variations of the SNR shown in FIG.


18


A. However, as described above, since the power correction information P


n,adj


(t) is created, and power adjustment is performed during decoding, the separation of the left and right signals can also be improved. In

FIGS. 18B

,


18


C, and


18


D, lines L


1


, L


3


, and L


5


indicate the characteristics of the coding device


91


of

FIG. 8

, and lines L


2


, L


4


, and L


6


indicate the characteristics of a prior coding device.




In the above-described embodiment of the present invention, the distortion factor of coding is detected, a mixing coefficient is set according to that value, and the input signal of the next timing is mixed. In addition, the construction may be formed in such a way that the input signal of a predetermined band is repeatedly mixed until the distortion factor becomes equal to or smaller than a predetermined threshold value. In this case, the signal L


n


(t) generated by the filter bank


101


-


1


and the signal R


n


(t) generated by the filter bank


101


-


2


are stored in a memory (not shown), etc., and mixing, domain conversion, and coding are performed again on the basis of the distortion factor information E


n


(f) which is fed back to the adaptive mixing section


102


.





FIG. 19

is a block diagram showing an example of the construction of a decoding device to which the present invention is applied.




A demultiplexer


161


divides the code sequence C supplied via a transmission line (not shown) into code sequences C


n


, C


n−1


, . . . , C


1


for each predetermined band, and outputs each code sequence C


i


to a corresponding decoding section (for the sake of convenience of description, only a decoding section


162


is shown). The code sequence C


n


is supplied to the decoding section


162


.




The decoding section


162


decodes the input code sequence C


n


by a decoding method corresponding to the coding method, outputs the obtained spectrum signal L′


n


(f) to a domain conversion section


163


-


1


, and outputs the obtained spectrum signal R′


n


(f) to a domain conversion section


163


-


2


. Furthermore, the decoding section


162


supplies the power correction information P


n,adj


(t) obtained from the code sequence C


n


to a power weighting section


164


.




The domain conversion section


163


converts the input spectrum signals L′


n


(f) and R′


n


(f) into signals of the time domain by using inverse MDCT, etc., and outputs the obtained signals L′


n


(t)


mix


and R′


n


(t)


mix


to a power weighting section


164


.




The power weighting section


164


performs power correction on the signals L′n(t)


mix


and R′n(t)


mix


supplied from the domain conversion section


163


on the basis of the power weighting coefficient contained in the supplied power correction information P


n,adj


(t), and outputs the generated signal L′n(t) to a filter bank


165


-


1


and outputs the generated signal R′n(t) to a filter bank


165


-


2


.




The filter bank


165


combines the signals L′n(t) and R′n(t) supplied from the power weighting section


164


with the signals L′


n−1


(t), . . . , L′


1


(t) and R′


n−1


(t), . . . , R′


1


(t) of the other bands, and outputs the generated audio signals L′(t) and R′(t) of all the bands to outside the decoding device


151


.





FIG. 20

is a block diagram showing a detailed example of the construction of the power weighting section


164


.




A power weighting coefficient setting section


171


sets a power weighting coefficient c contained in the supplied power correction information P


n,adj


(t) in a multiplier


172


-


1


and sets a power weighting coefficient d in a multiplier


172


-


2


.




The multiplier


172


-


1


multiplies the input signal L′


n


(t)


mix


by the power weighting coefficient c. The multiplier


172


-


2


multiplies the input signal R′


n


(t)


mix


by the power weighting coefficient d. The obtained signals L′


n


(t) and R′


n


(t) are output to the filter banks


165


-


1


and


165


-


2


, respectively.





FIG. 21

is a block diagram showing a detailed example of the construction of the multiplier


172


-


1


(although not shown, the multiplier


172


-


2


is also similarly constructed).




In this example, buffers


172


A and


172


B are provided. At the current time (time t=0), the set power weighting coefficient c(t


0


) is stored in the buffer


172


A, and the power weighting coefficient c(t


1


) which was set immediately before (which has been set at time t=1) is stored in the buffer


172


B.




More specifically, when the power weighting coefficient c(t) is changed, there are cases in which a noncontinuous point occurs in the signal output at that time. Therefore, as indicated in lines i to iii of

FIG. 22

, the occurrence of a noncontinuous point can be prevented by changing the power weighting coefficient c(t) in a manner of a straight line or in a manner of a curve. Although in this example, two buffers are provided, three or more buffers may be provided. A degree of the interpolation function which interpolates each power weighting coefficient may be one, two, three, etc.




Next, referring to the flowchart in

FIG. 23

, the decoding process of the decoding device


151


of

FIG. 19

will be described.




In step S


71


, the demultiplexer


161


divides the input code sequence C to code sequences C


n


, C


n−1


, . . . , C


1


of a predetermined number of bands n, and outputs them to the corresponding decoding sections.




In step S


72


, the decoding section


162


selects a decoding method on the basis of a combination of normalization information, quantization accuracy information, a codebook number, etc., decodes the input code sequence C


n


, outputs the obtained spectrum signal L′n(f) to the domain conversion section


163


-


1


, and outputs the spectrum signal R′n(f) to the domain conversion section


163


-


2


. Furthermore, the decoding section


162


outputs the power correction information P


n,adj


(t) obtained from the code sequence C


n


to the power weighting section


164


.




In step S


73


, the domain conversion sections


163


-


1


and


163


-


2


convert the input spectral signals L′


n


(f) and R′


n


(f) into the signals in the time domain by using inverse MDCT, etc., and outputs the obtained signals L′


n


(t)


mix


and R′


n


(t)


mix


to the power weighting section


164


. The signals L′


n


(t)


mix


and R′


n


(t)


mix


are signals having a possibility that mixing was performed in the coding device


91


, and there are cases in which the originally stereo signal is changed to a substantially monaural signal. Therefore, in step S


74


, the power weighting section


164


performs a power weighting process on the basis of the supplied power correction information P


n,adj


(t), thereby reproducing a pseudo-stereo signal. The details of the power weighting process will be described later with reference to the flowchart in FIG.


24


.




The signals L′


n


(t) and R′


n


(t) obtained by the power weighting process are output to the filter banks


165


-


1


and


165


-


2


, respectively. The above process is performed for each band.




Then, in step S


75


, the filter bank


165


combines the signals L′


n


(t) and R′


n


(t) supplied from the power weighting section


164


with the signals L′


n−1


(t) L′


1


(t), R′


n−1


(t), . . . , R′


1


(t) of the other bands, and outputs the combined audio signals L′


n


(t) and R′


n


(t) of all the bands to outside the decoding device


151


.




Next, referring to the flowchart in

FIG. 24

, the power weighting process performed in step S


74


of

FIG. 23

will be described.




In step S


91


, the power weighting coefficient setting section


171


sets the power weighting coefficients c and d of the multipliers


172


-


1


and


172


-


2


on the basis of the power weighting coefficient contained in the power correction information P


n,adj


(t) supplied from the decoding section


162


.




In step S


92


, the multipliers


172


-


1


and


172


-


2


multiply the input signals L′


n


(t)


mix


and R′


n


(t)


mix


by the power weighting coefficients c and d, respectively, and outputs the generated signals L′


n


(t) and R′


n


(t) to the filter banks


165


-


1


and


165


-


2


, respectively.




For example, as described above, in a case where the power correction information P


n,adj


(t) (power weighting coefficients) is set as c=1.25 and d=0.05 in the power correction section


123


, and the respective power weighting coefficients c and d are set by the power weighting coefficient setting section


171


, the multiplier


172


-


1


multiplies the power of the input signal L′


n


(t)


mix


by 1.25, and outputs the generated signal L′n(t) to the filter bank


165


-


1


. Also, the multiplier


172


-


2


multiplies the power of the input signal R′


n


(t)


mix


by 0.05, and outputs the generated signal R′n(t) to the filter bank


165


-


2


.




As a result, even when the separation of the left and right signals become poorer by coding, it is possible to reproduce a pseudo-stereo signal.




Although the above-described series of processes can be performed by hardware, it can also be performed by software. In this case, for example, the coding device


91


is formed of a personal computer


181


such as that shown in FIG.


25


.




In

FIG. 25

, a CPU (Central Processing Unit)


191


performs various processing in accordance with a program stored in a ROM (Read Only Memory)


192


or a program loaded into a RAM (Random Access Memory)


193


from a storage section


198


. Also, in the RAM


193


, data, etc., required when the CPU


191


performs various processing is stored as appropriate.




The CPU


191


, the ROM


192


, and the RAM


193


are interconnected with each other via a bus


194


. An input/output interface


195


is also connected to this bus


194


.




An input section


196


including a keyboard, a mouse, etc., an output section


197


including a display formed of a CRT or an LCD (Liquid-Crystal Display), a speaker, etc., a storage section


198


formed of a hard disk, etc., and a communication section


199


formed of a modem, a terminal adapter, etc., are connected to the input/output interface


195


. The communication section


199


performs a communication process via a network.




A drive


200


is also connected to the input/output interface


195


as necessary. A magnetic disk


201


, an optical disk


202


, a magneto-optical disk


203


, a semiconductor memory


204


, etc., is loaded into the drive


200


where appropriate, and a computer program read therefrom is installed into the storage section


198


as necessary.




In a case where a series of processes is performed by software, programs which form the software are installed from a network or a recording medium into a computer incorporated into dedicated hardware or into, for example, a general-purpose personal computer


181


, etc., capable of executing various types of functions by installing various programs.




This recording medium, as shown in

FIG. 25

, is constructed by not only package media formed of the magnetic disk


201


(including a floppy disk), the optical disk


202


(including a CD-ROM, and a DVD (Digital Versatile Disk)), the magneto-optical disk


203


(including an MD (Mini-Disk)), or the semiconductor memory


204


, in which programs are recorded, which is distributed separately from the main unit of the device so as to distribute programs to a user, but also is constructed by the ROM


192


, a hard disk contained in the storage section


198


, etc., in which programs are recorded, which is distributed to a user in a state in which it is incorporated in advance into the main unit of the device.




In this specification, steps which describe a program recorded in a recording medium contain not only processing performed in a time-series manner along the described sequence, but also processing performed in parallel or individually although the processing is not necessarily performed in a time-series manner.




While the present invention has been described with reference to what are presently considered to be the preferred embodiments, it is to be understood that the invention is not limited to the disclosed embodiments. On the contrary, the invention is intended to cover various modifications and equivalent arrangements included within the spirit and scope of the appended claims. The scope of the following claims is to be accorded the broadest interpretation so as to encompass all such modifications and equivalent structures and functions.



Claims
  • 1. A coding device for coding an input signal, comprising:coding method selection means for selecting a coding method in accordance with the input signal; coding means for coding said input signal in accordance with said coding method selected by said coding method selection means; distortion factor detection means for detecting a distortion factor of coding by said coding means; and mixing means for mixing left and right components of said input signal on the basis of a mixing ratio determined in such a manner as to correspond to said distortion factor detected by said distortion factor detection means, wherein said coding method selection means selects said coding method in accordance with said input signal mixed by said mixing means.
  • 2. A coding device according to claim 1, further comprising output correction information creation means for creating output correction information which is used when said input signal coded by said coding means is decoded.
  • 3. A coding device according to claim 1, wherein said coding method selection means selects said coding method for use with said input signal on the basis of a threshold value determined according to the construction of said coding device.
  • 4. A coding device for coding an input signal, comprising:coding method selection means for selecting a coding method in accordance with the input signal; coding means for coding said input sianal in accordance with said coding method selected by said coding method selection means; distortion factor detection means for detecting a distortion factor of coding by said coding means; and mixing means for mixing left and right components of said input signal on the basis of a mixing ratio determined in such a manner as to correspond to said distortion factor detectcd by said distortion factor detection means, wherein, said coding method selection means selects said coding method in accordance with said input signal mixed by said mixing means, and said coding method said coding method selection means selects said coding method from among a dual coding method, an intermediate portion/side portion stereo coding method, and an intensity stereo coding method.
  • 5. A coding device according to claim 4, wherein said coding method selection means selects said dual coding method when the correlation of the left and right components of said input signal is low.
  • 6. A coding device according to claim 5, wherein said coding method selection means determines the correlation of the left and right components of said input signal by using the ratio of the total sum of the sum signals with respect to the total sum of difference signals of said left and right components.
  • 7. A coding device according to claim 5, wherein, when the correlation of the left and right components of said input signal is high, said coding method selection means determines which one of the MS stereo coding and the IS stereo coding should be selected on the basis of a maximum value of the difference signal of said left and right components.
  • 8. A coding device for coding an input signal, comprising:coding method selection means for selecting a coding method in accordance with the input signal; coding means for coding said input signal in accordance with said coding method selected by said coding method selection means; distortion factor detection means for detecting a distortion factor of coding by said coding means; and mixing means for mixing left and right components of said input signal on the basis of a mixing ratio determined in such a manner as to correspond to said distortion factor detected by said distortion factor detection means, wherein, said coding method selection means selects said coding method in accordance with said input signal mixed by said mixing means, and said mixing means stores said mixing ratio, and changes said mixing ratio on the basis of an interpolation function of said mixing ratio determined immediately before and said mixing ratio determined currently.
  • 9. A coding device according to claim 1, further comprising input signal storage means for storing said input signal, wherein said mixing means mixes the left and right components of the particular input signal stored in said input signal storage means at least once on the basis of the mixing ratio determined in such a manner as to correspond to the distortion factor when the particular input signal was coded.
  • 10. A coding method for coding an input signal, comprising:a coding method selection step of selecting a coding method in accordance with the input signal; a coding step of coding said input signal in accordance with said coding method selected in said coding method selection step; a distortion factor detection step of detecting a distortion factor of coding in said coding step; and a mixing step of mixing the left and right components of said input signal on the basis of a mixing ratio determined in such a manner as to correspond to said distortion factor detected in said distortion factor detection step, wherein the process of said coding method selection step selects said coding method in accordance with said input signal mixed in said mixing step.
  • 11. A recording medium having recorded thereon a computer-readable program, said program comprising:a coding method selection step of selecting a coding method in accordance with an input signal; a coding step of coding said input signal in accordance with said coding method selected in said coding method selection step; a distortion factor detection step of detecting a distortion factor of coding in said coding step; and a mixing step of mixing the left and right components of said input signal on the basis of a mixing ratio determined in such a manner as to correspond to said distortion factor detected in said distortion factor detection step, wherein the process of said coding method selection step selects said coding method in accordance with said input signal mixed in said mixing step.
  • 12. A decoding device for decoding a code sequence coded by a predetermined coding method, said decoding device comprising:decoding method selection means for selecting a decoding method corresponding to said coding method; decoding means for decoding an input code sequence in accordance with said decoding method selected by said decoding method selection means; correction means for correcting the left and right components of a signal decoded by said decoding means on the basis of information supplied from said coding device; and output means for outputting said signal corrected by said correction means.
  • 13. A decoding method for decoding a code sequence coded by a predetermined coding method, said decoding method comprising:a decoding method selection step of selecting a decoding method corresponding to a coding method used by a coding device; a decoding step of decoding an input code sequence in accordance with said decoding method selected in said decoding method selection step; a correction step of correcting the left and right components of a signal decoded in said decoding step on the basis of information supplied from said coding device; and an output step of outputting said signal corrected in said correction step.
  • 14. A recording medium having recorded thereon a computer-readable program, said program comprising:a decoding method selection step of selecting a decoding method corresponding to a coding method used by a coding device; a decoding step of decoding an input code sequence in accordance with said decoding method selected in said decoding method selection step; a correction step of correcting the left and right components of a signal decoded in said decoding step on the basis of information supplied from said coding device; and an output step of outputting said signal corrected in said correction step.
Priority Claims (1)
Number Date Country Kind
P2000-380642 Dec 2000 JP
US Referenced Citations (2)
Number Name Date Kind
6240386 Thyssen et al. May 2001 B1
6356211 Shimoyoshi et al. Mar 2002 B1