Patent Document

CROSS-REFERENCE TO RELATED APPLICATION  
       [0001]     This application claims the benefit of Korean Patent Application No. 10-2005-0020136, filed on Mar. 10, 2005, in the Korean Intellectual Property Office, the disclosure of which incorporated herein by reference.  
       BACKGROUND OF THE INVENTION  
       [0002]     1. Field of the Invention  
         [0003]     The present invention relates to audio coding and decoding apparatuses and methods, and recording mediums on which the methods are recorded, and more particularly, to audio coding and decoding apparatuses and methods in which the quality of an audio signal including harmonics can be optimized, and recording mediums on which the methods are recorded.  
         [0004]     2. Description of Related Art  
         [0005]     As the range of applications of audio coders has increased, the demand for low transmission rate coders has also increased. As such, a code excited linear prediction (CELP) coder is being used for transmission rates equal to or greater than 4 kbps, and a harmonic-CELP coder is being used for transmission rates of less than 4 kbps. The reason why a harmonic-CELP coder is being used for transmission rates of less than 4 kbps is that, in a CELP coding algorithm, sound quality is lowered when there are too few quantization bits, whereas, in a harmonic coding algorithm, the periodicity of a voiced sound that greatly affects sound quality, even fewer smaller bits, is well modeled.  
         [0006]     A harmonic vector excitation coder (HVXC), which uses the MPEG-4 audio standard is an example of a harmonic-CELP coder. An HVXC is characterized by quantization of a variable dimension harmonic vector, high-speed harmonic synthesis, harmonic amplitude estimation using a real number pitch, and natural property control using noise mixing.  
         [0007]     However, in a harmonic-CELP coder, an audio signal section (or voiced sound section) including harmonics is formed by interpolating standard waveforms of a previous frame and a current frame so that there is a high probability that pitch halving prediction in which a pitch lag is reduced by half or pitch doubling prediction in which a pitch lag is doubled can be performed in a transition section of the harmonic-CELP coder. When the pitch halving prediction or the pitch doubling prediction is performed, waveform distortion and discontinuity occur at a frame boundary due to a severe amount of variation of pitch lag.  
         [0008]     In addition, since an overlap-addition method through a triangular window is used in harmonic synthesis, when a signal in an audio signal section including harmonics in a transition section increases or decreases instantaneously, a synthesis excitation signal may disadvantageously increase or decrease linearly due to the effect of the triangular window.  
       BRIEF SUMMARY  
       [0009]     An aspect of the present invention provides audio coding and decoding apparatuses and methods in which the quality of an audio signal including harmonics can be optimized, and recording mediums on which the methods are recorded.  
         [0010]     An aspect of the present invention also provides audio coding and decoding apparatuses and methods in which pitch halving prediction or pitch doubling prediction in an audio signal section including harmonics can be prevented, and recording mediums on which the methods are recorded.  
         [0011]     An aspect of the present invention also provides audio coding and decoding apparatuses and methods in which harmonic amplitude information is converted into a quantized LPC coefficient and the quantized LPC coefficient is used to extract LPC coefficients needed by a second harmonic coding module and a CELP module, and recording mediums on which the methods are recorded.  
         [0012]     An aspect of the present invention also provides audio coding and decoding apparatuses and methods in which bit allocation for a plurality of coding modules is performed differently according to whether harmonics are included in an input audio signal, and recording mediums on which the methods are recorded.  
         [0013]     An aspect of the present invention also provides audio coding and decoding apparatuses and methods in which scalability can be easily applied, and recording mediums on which the methods are recorded.  
         [0014]     According to an aspect of the present invention, there is provided an audio coding apparatus, the audio coding apparatus including: a first harmonic coding module performing first harmonic coding on an input audio signal using a pitch lag of the input audio signal and producing a quantized linear prediction coding coefficient; a first detector detecting a first difference audio signal from a difference between an audio signal output from the first harmonic coding module and the input audio signal; a second harmonic coding module performing harmonic coding on the first difference audio signal using the quantized linear prediction coding coefficient and a previous harmonic coding result; a second detector detecting a second difference audio signal obtained from a difference between an audio signal output from the second harmonic coding module and the first difference audio signal; and a code excited linear prediction (CELP) module CELP coding the second difference audio signal using the quantized linear prediction coding coefficient obtained from the first harmonic coding module.  
         [0015]     The first harmonic coding module may convert an amplitude of harmonics of the input audio signal into a linear prediction coding coefficient, quantize the converted linear prediction coding coefficient, and provide the quantized linear prediction coding coefficient to the second harmonic coding module and the CELP module, respectively.  
         [0016]     The second harmonic coding module may extract a quantized linear prediction coding coefficient needed for the second harmonic coding using the quantized linear prediction coding coefficient obtained from the first harmonic coding module.  
         [0017]     According to another aspect of the present invention, there is provided an audio decoding apparatus, the audio decoding apparatus including: an inverse quantization unit inverse quantizing each of a plurality of parameters to restore an audio signal; a first harmonic decoding module performing harmonic decoding using a linear prediction coding coefficient and a phase vector output from the inverse quantization unit; a second harmonic decoding module performing harmonic decoding based on the linear prediction coding coefficient, a harmonic index, and a first gain value output from the inverse quantization unit; a first adder adding a signal output from the first harmonic decoding module to a signal output from the second harmonic decoding module; a code excited linear prediction (CELP) decoding module performing CELP decoding based on a stochastic codebook index output from the inverse quantization unit and a second gain value output from the inverse quantization unit; and a second adder adding a signal output from the first adder to a signal output from the CELP decoding module and outputting the result as a restored audio signal.  
         [0018]     According to another aspect of the present invention, there is provided an audio coding method, the audio coding method including: harmonically coding an input audio signal without analyzing a linear prediction coding coefficient; analyzing a linear prediction coding coefficient of a difference audio signal obtained from a difference between the input audio signal and the harmonic-coding result and harmonically coding the difference audio signal; and CELP coding a difference audio signal obtained from a difference between the result of harmonically coding on the difference audio signal and the input audio signal.  
         [0019]     According to another aspect of the present invention, there is provided an audio decoding method, the audio decoding method including: inverse quantizing a plurality of parameters for restoring an audio signal; first harmonic decoding using a linear prediction coding coefficient and a phase vector obtained through the inverse quantizing; second harmonic decoding using a linear prediction coding coefficient, a harmonic index, and a first gain value obtained through the inverse quantizing; first adding the first harmonic decoding result to the second harmonic decoding result; CELP decoding using a stochastic index and a second gain value obtained through the inverse quantization; and adding the result obtained through the first adding to the result obtained through the CELP decoding to obtain a restored audio signal.  
         [0020]     According to another aspect of the present invention, there is provided a recording medium on which a program for performing an audio coding method is recorded, the audio coding method including: harmonically coding an input audio signal without analyzing a linear prediction coding coefficient; analyzing a linear prediction coding coefficient of a difference audio signal obtained from a difference between the input audio signal and the harmonic-coding result and harmonically coding the difference audio signal; and CELP coding a difference audio signal obtained from a difference between the result of harmonically coding on the difference audio signal and the input audio signal.  
         [0021]     Additional and/or other aspects and advantages of the present invention will be set forth in part in the description which follows and, in part, will be obvious from the description, or may be learned by practice of the invention.  
     
    
     BRIEF DESCRIPTION OF THE DRAWINGS  
       [0022]     The above and/or other aspects and advantages of the present invention will become apparent and more readily appreciated from the following detailed description, taken in conjunction with the accompanying drawings of which:  
         [0023]      FIG. 1  is a functional block diagram of an audio coding apparatus according to an embodiment of the present invention;  
         [0024]      FIG. 2  is a detailed block diagram of a first harmonic coding module shown in  FIG. 1 ;  
         [0025]      FIG. 3  is a detailed block diagram of a second harmonic coding module shown in  FIG. 1 ;  
         [0026]      FIG. 4  is a detailed block diagram of a CELP module shown in  FIG. 1 ;  
         [0027]      FIG. 5  is a functional block diagram of an audio decoding apparatus according to another embodiment of the present invention;  
         [0028]      FIG. 6  is a flowchart illustrating an audio coding method according to another embodiment of the present invention; and  
         [0029]      FIG. 7  is a flowchart illustrating an audio decoding method according to another embodiment of the present invention. 
     
    
     DETAILED DESCRIPTION OF EMBODIMENTS  
       [0030]     Reference will now be made in detail to embodiments of the present invention, examples of which are illustrated in the accompanying drawings, wherein like reference numerals refer to the like elements throughout. The embodiments are described below in order to explain the present invention by referring to the figures.  
         [0031]      FIG. 1  is a functional block diagram of an audio coding apparatus according to an embodiment of the present invention. Referring to  FIG. 1 , the audio coding apparatus includes a pitch analyzer  110 , a signal classifier  120 , a bit allocator  130 , a first harmonic coding module  140 , a first detector  150 , a second harmonic coding module  160 , a second detector  170 , and a code excited linear prediction (CELP) module  180 .  
         [0032]     The pitch analyzer  110  analyzes the pitch of an input audio signal and detects a pitch lag t p . The pitch lag t p  is obtained using a normalized auto-correlation function shown in Equation 1 
                 R   ⁡     (   i   )       =         ∑     n   =   0         L   I     -   1       ⁢       s   ⁡     (   n   )       ⁢     s   ⁡     (     n   -   i     )                 ∑     n   =   0         L   I     -   1       ⁢       s   2     ⁡     (     n   -   i     )               ,     i   =     L   MIN       ,   …   ⁢           ,     L   MAX     ,           (   1   )             
 
         [0033]     where s(n) is the input audio signal, L f  is the length of a portion of the audio signal s(n) to be analyzed, and L MIN  and L MAX  are the maximum and minimum of the pitch, respectively. In general, L MIN  and L MAX  are 20 and 143, respectively. Maximum values of R(i) are found for L MIN   ≮   ≮   MIN +19, L MIN +20 ≮   ≮   MIN +39, L MIN +40 ≮   ≮   MAX , respectively. If the respective values of i as t 3 , t 2 , and t 1 , one value is selected from t 3 , t 2 , and t 1  as a pitch lag t p  based on Equation 2.  
                                                        t p =t 1                 R(t p ) = R(t 1 )              if R(t 2 ) = 0.85 R(t p )                R(t p ) = R(t 2 )                 t p  = t 2                  End              if R(t 3 ) = 0.85R(t p )                R(t p ) = R(t 3 )                 t p  = t 3             end                (2)                      
 
         [0034]     The pitch lag t p  detected by the pitch analyzer  110  is provided to the first harmonic coding module  140 .  
         [0035]     The signal classifier  120  determines whether harmonics are included in the input audio signal. That is, the signal classifier  120  detects values of the input signal such as a sharpness rate, a right and left energy rate, a zero-crossing rate, and a first-order prediction coefficient, compares a threshold value for each detected value with the detected values, and if the comparison result satisfies a predetermined condition, the signal classifier  120  can determine that the harmonics are included in the input audio signal. The comparison can be performed in subframe units. The determination result of the signal classifier  120  is provided to the bit allocator  130 .  
         [0036]     The bit allocator  130  provides allocation bit information for the first harmonic coding module  140 , the second harmonic coding module  150 , and the CELP module  180  according to the determined result provided by the signal classifier  120 . If a signal indicating that the harmonics are included in the input audio signal is provided by the signal classifier  120 , the bit allocator  130  can provide information indicating that bits are allocated at a ratio of 3:3:2, for example, to the first harmonic coding module  140 , the second harmonic coding module  150 , and the CELP module  180 . If a signal indicating that the harmonics are not included in the input audio signal is provided by the signal classifier  120 , the bit allocator  130  can provide information indicating that bits are allocated at a ratio of 2:2:4, for example, to the first harmonic coding module  140 , the second harmonic coding module  150 , and the CELP module  180 . The bit allocation information can be set in advance.  
         [0037]     The first harmonic coding module  140  performs harmonic coding on the input audio signal using the pitch lag and outputs a linear prediction coding (LPC) coefficient quantized for audio decoding, a quantized LPC (QLPC) coefficient index, and a quantized phase index.  
         [0038]     To this end, the first harmonic coding module  140  includes a first harmonic analyzer  201 , an amplitude/LPC coefficient converter  202 , an LPC coefficient quantizer  203 , a QLPC/amplitude converter  204 , a phase quantizer  205 , and a first harmonic synthesizer  206 , as shown in  FIG. 2 .  
         [0039]     The first harmonic analyzer  201  analyzes harmonics of the input audio signal using a pitch lag (or a pitch delay). That is, the first harmonic analyzer  201  searches for a fundamental frequency ω 0  using the pitch lag and searches for harmonic parameters using a sine dictionary. The harmonic parameters include an amplitude A and a phase φ.  
         [0040]     The amplitude A and the phase φ of the sine dictionary are found using a matching pursuit (MP) algorithm in which the input audio signal s(n) is used as a target signal. The input audio signal S H (n) can be expressed using the sine dictionary as shown in Equation 3 
                   s   H     ⁡     (   n   )       =         w   ham     ⁡     (   n   )       ⁢       ∑     k   =   0       K   -   1       ⁢       A   k     ⁢   cos   ⁢           ⁢     (         ω   k     ⁢   n     +     ϕ   k       )             ,           (   3   )             
 
 where A k  is the amplitude of a k-th sine wave, ω k  is an angle frequency of the k-th sine wave, φ k  is the phase of the k-th sine wave, w ham (n) is a hamming window, and K is the number of sine dictionaries, which is generally obtained using Equation 4.  
             K   =     ⌊       t   p     2     ⌋             (   4   )             
 
         [0041]     The angle frequency ω k  of sine dictionaries can be obtained using Equation 5.  
               ω   k     =         2   ⁢           ⁢   π       t   p       ⁢     (     k   +   1     )               (   5   )             
 
         [0042]     Referring to  FIGS. 1 and 2 , the search for the amplitude A and the phase φ of the sine dictionary using the MP algorithm is performed in such a way that an operation of projecting a k-th target signal on a k-th sine dictionary to extract the amplitude of a component and an operation of offsetting the extracted amplitude of the component with the k-th target signal to generate a new (k+1)-th target signal are repeatedly performed. The amplitude and the phase of the sine dictionary using the MP algorithm can be found using Equation 6 
                 E   k     =       ∑     n   =   0         L   sf     -   1       ⁢           w   ham     ⁡     (   n   )       ⁡     [         r     h   ,   k       ⁡     (   n   )       -       A   k     ⁢   cos   ⁢           ⁢     (         ω   k     ⁢   n     +     ϕ   k       )         ]       2         ,           (   6   )             
 
 where r h,k  is a k-th target signal and E k  is a value obtained by multiplying a mean squared error between r h,k  and a k-th sine dictionary by a hamming window wham. If k=0, r h,k (n) is the same as the original audio signal s(n). A k  and φ k  which minimize E k  can be defined using Equation 7.  
                   A   k     =         a   k   2     +     b   k   2           ,       ϕ   k     =     -       tan     -   1       ⁡     (       b   k       a   k       )             ⁢     
     ⁢         a   k     =                 ∑     n   =   0         L   sf     -   1       ⁢         sin   2     ⁡     (       ω   k     ⁢   n     )       ⁢       ∑     n   ⁢           =           ⁢   0               ⁢       L             ⁢   sf       ⁢           -           ⁢   1         ⁢         r     h   ,   k       ⁡     (   n   )       ⁢   cos   ⁢           ⁢     (       ω   k     ⁢   n     )             -                 ∑     n   =   0         L   sf     -   1       ⁢       cos   ⁡     (       ω   k     ⁢   n     )       ⁢     sin   ⁡     (       ω   k     ⁢   n     )       ⁢       ∑     n   ⁢           =           ⁢   0               ⁢       L             ⁢   sf       ⁢           -           ⁢   1         ⁢         r     h   ,   k       ⁡     (   n   )       ⁢   sin   ⁢           ⁢     (       ω   k     ⁢   n     )                               ∑     n   =   0         L   sf     -   1       ⁢         cos   2     ⁡     (       ω   k     ⁢   n     )       ⁢       ∑     n   =   0         L   sf     -   1       ⁢       sin   2     ⁡     (       ω   k     ⁢   n     )             -                 ∑     n   =   0         L   sf     -   1       ⁢     cos   ⁢           ⁢     (       ω   k     ⁢   n     )     ⁢           ⁢   sin   ⁢           ⁢     (       ω   k     ⁢   n     )     ⁢       ∑     n   =   0         L   sf     -   1       ⁢     cos   ⁢           ⁢     (       ω   k     ⁢   n     )     ⁢     sin   ⁡     (       ω   k     ⁢   n     )                         ,     
     ⁢       b   k     =                 ∑     n   =   0         L   sf     -   1       ⁢         cos   2     ⁡     (       ω   k     ⁢   n     )       ⁢       ∑     n   ⁢           =           ⁢   0               ⁢       L             ⁢   sf       ⁢           -           ⁢   1         ⁢         r     h   ,   k       ⁡     (   n   )       ⁢   sin   ⁢           ⁢     (       ω   k     ⁢   n     )             -                 ∑     n   =   0         L   sf     -   1       ⁢       cos   ⁡     (       ω   k     ⁢   n     )       ⁢     sin   ⁡     (       ω   k     ⁢   n     )       ⁢       ∑     n   ⁢           =           ⁢   0               ⁢       L             ⁢   sf       ⁢           -           ⁢   1         ⁢         r     h   ,   k       ⁡     (   n   )       ⁢   cos   ⁢           ⁢     (       ω   k     ⁢   n     )                               ∑     n   =   0         L   sf     -   1       ⁢         cos   2     ⁡     (       ω   k     ⁢   n     )       ⁢       ∑     n   =   0         L   sf     -   1       ⁢       sin   2     ⁡     (       ω   k     ⁢   n     )             -                 ∑     n   =   0         L   sf     -   1       ⁢     cos   ⁢           ⁢     (       ω   k     ⁢   n     )     ⁢           ⁢   sin   ⁢           ⁢     (       ω   k     ⁢   n     )     ⁢       ∑     n   =   0         L   sf     -   1       ⁢     cos   ⁢           ⁢     (       ω   k     ⁢   n     )     ⁢     sin   ⁡     (       ω   k     ⁢   n     )                                   (   7   )             
 
         [0043]     The first harmonic analyzer  201  transmits the amplitude of the sine dictionary to the amplitude/LPC coefficient converter  202  and transmits the phase of the sine dictionary to the phase quantizer  205 .  
         [0044]     The amplitude/LPC coefficient converter  202  converts the amplitude A of the input sine dictionary into an LPC coefficient. The LPC coefficient analyzer  203  quantizes the LPC coefficient using the allocated bit information provided by the bit allocator  130  and outputs the quantized LPC (QLPC) coefficient and the quantized LPC coefficient index.  
         [0045]     The QLPC/amplitude converter  204  converts the quantized LPC coefficient into an amplitude vector Â of the quantized sine dictionary and outputs the amplitude vector Â.  
         [0046]     The phase quantizer  205  quantizes a phase output from the first harmonic analyzer  201  based on the allocated bit information provided by the bit allocator  130  and outputs a quantized phase vector {circumflex over (φ)} and a quantized phase index.  
         [0047]     The first harmonic synthesizer  206  synthesizes the amplitude vector Â of the quantized sine dictionary output from the QLPC/amplitude converter  204  and the quantized phase vector {circumflex over (φ)} output from the phase quantizer  205  using Equation 8 to obtain a synthesized audio signal {circumflex over (S H )}(n) with respect to the input audio signal.  
                   s   ^     H     ⁡     (   n   )       =       w   ham     ⁢       ∑     k   =   0     K     ⁢         A   ^     k     ⁢   cos   ⁢           ⁢     (         ω   k     ⁢   n     +       ϕ   ^     k       )                   (   8   )             
 
         [0048]     The first harmonic synthesizer  206  transmits the synthesized audio signal {circumflex over (s H )}(n) to the first detector  150 .  
         [0049]     The first detector  150  detects and outputs a first difference audio signal obtained from the first difference between the input audio signal and the synthesized audio signal output from the first harmonic coding module  140 .  
         [0050]     The second harmonic coding module  160  harmonically codes the first difference audio signal detected by the first detector  150  using the quantized LPC coefficient obtained by the first harmonic coding module  140  and a previous output signal of the second harmonic coding module  160 , outputs a first synthesized difference audio signal, a harmonic index quantized for audio signal decoding and a first quantized gain index.  
         [0051]     To this end, referring to  FIG. 3 , the second harmonic coding module  160  includes an LPC coefficient analyzer  301 , an inverse synthesis filter  302 , a second harmonic analyzer  303 , an index quantizer  304 , a second harmonic synthesizer  305 , and a synthesis filter  306 .  
         [0052]     The LPC coefficient analyzer  301  analyzes an LPC coefficient on the first difference audio signal output from the first detector  150  using the quantized LPC coefficient provided by the first harmonic coding module  140  and extracts an LPC coefficient needed by the second harmonic coding module  160 .  
         [0053]     The LPC coefficient analyzer  301  can be configured to extract a reduced LPC coefficient when the order of the quantized LPC coefficient provided by the first harmonic coding module  140  must be reduced according to the operation conditions of a corresponding audio coding apparatus. An LPC coefficient can be reduced by obtaining only necessary LPC coefficients in a head part among transmitted LPC coefficients. In this case, the number of LPC coefficients should be even. For example, when the order of the quantized LPC coefficient is P and the order of an LPC coefficient to be intended to be used in the second harmonic coding module  160  is Q, the number of Q LPC coefficients existed in the head part are extracted from all P LPC coefficients. The extracted LPC coefficients are provided to the inverse synthesis filter  302  and the synthesis filter  306 , respectively.  
         [0054]     The inverse synthesis filter  302  performs the inverse operation of the operation performed by a synthesis filter on the first difference audio signal detected by the first detector  150  to generate an excitation signal of the first difference audio signal and transmits the generated excitation signal to the second harmonic analyzer  303 .  
         [0055]     Referring to  FIGS. 1-3 , the second harmonic analyzer  303  has the same structure as the first harmonic analyzer  201  of  FIG. 2 , searches for an amplitude A and a phase φ of the sine dictionary with respect to the excitation signal output from the inverse synthesis filter  302  and outputs a harmonic index including the amplitude A and the phase φ of the sine dictionary. The output harmonic index is transmitted to the index quantizer  304 .  
         [0056]     The index quantizer  304  quantizes the harmonic index output from the second harmonic analyzer  303  using the allocated bit information provided by the bit allocator  130  and outputs the quantized harmonic index and the quantized gain index.  
         [0057]     The second harmonic synthesizer  305  has the same structure as the first harmonic synthesizer  206  of  FIG. 2 , synthesizes the quantized harmonic index output from the index quantizer  304 , and outputs the synthesized audio signal.  
         [0058]     The synthesis filter  306  outputs the first synthesized difference audio signal by synthesis filtering the synthesized audio signal output from the second harmonic synthesizer  305  using the quantized LPC coefficient output from the LPC coefficient analyzer  301 . The first synthesized difference audio signal is output to the second detector  170 .  
         [0059]     The second detector  170  detects a difference audio signal obtained from the difference between the first difference audio signal output from the first detector  150  and the first synthesized difference audio signal output from the second harmonic coding module  160  and outputs the detected difference audio signal as a second difference audio signal.  
         [0060]     The CELP module  180  CELP-codes the second difference audio signal output from the second detector  170  using the quantized LPC coefficient obtained by the first harmonic coding module  140  and outputs a stochastic index quantized and a second quantized gain index in order to decode an audio signal.  
         [0061]     To this end, the CELP module  180  includes a third detector  401 , a perceptual weighting filter  402 , a stochastic codebook search unit  403 , an index quantizer  404 , a stochastic codebook  405 , a multiplier  406 , an LPC coefficient analyzer  407 , and a synthesis filter  408 , as shown in  FIG. 4 .  
         [0062]     The third detector  401  detects a difference audio signal obtained from a difference between the second difference audio signal output from the second detector  170  and a synthesized audio signal previously obtained by the CELP module  180 .  
         [0063]     The perceptual weighting filter  402  perceptual-weighting-filters the difference audio signal using the LPC coefficient provided by the LPC coefficient analyzer  407  so that quantization noise of the difference audio signal output from the third detector  401  is equal to or less than a masking level using a hearing masking effect.  
         [0064]     The stochastic codebook search unit  403  searches one corresponding stochastic codebook based on a signal output from the perceptual weighting filter  402  and outputs an index of the searched stochastic codebook.  
         [0065]     The index quantizer  404  quantizes the index provided by the stochastic codebook search unit  403  and outputs the quantized stochastic codebook index and the quantized gain index.  
         [0066]     The stochastic codebook  405  includes a plurality of stochastic codebooks and outputs a stochastic codebook that corresponds to the quantized stochastic codebook index provided by the index quantizer  404 .  
         [0067]     The multiplier  406  multiplies the stochastic codebook output from the stochastic codebook  405  by the quantized gain output from the index quantizer  404 .  
         [0068]     The LPC coefficient analyzer  407  analyzes the quantized LPC coefficient of the signal output from the third detector  401  using the quantized LPC coefficient provided by the first harmonic coding module  140  and extracts the quantized LPC coefficient. The method of extracting the quantized LPC coefficient is similar to the method used in the LPC coefficient analyzer  301  provided in the second harmonic coding module  160 .  
         [0069]     The extracted LPC coefficient is provided to the perceptual weighting filter  402  and the synthesis filter  408 .  
         [0070]     The synthesis filter  408  performs synthesis filtering on the signal output from the multiplier  406  using the quantized LPC coefficient output from the LPC coefficient analyzer  407  and provides the synthesis-filtered result to the third detector  401 . The synthesis filtering is performed by obtaining an impulse response of the synthesis filter  408  from the quantized LPC coefficient and then convoluting the impulse response and the signal output from the multiplier  406  to obtain the synthesized audio signal.  
         [0071]      FIG. 5  is a functional block diagram of an audio decoding apparatus according to another embodiment of the present invention. Referring to  FIG. 5 , the audio decoding apparatus of  FIG. 5  includes an LPC coefficient inverse quantizer  501 , a phase index inverse quantizer  502 , a harmonic index inverse quantizer  503 , a first gain index inverse quantizer  504 , a stochastic index quantizer  505 , a second gain index inverse quantizer  506 , a first harmonic decoding module  510 , a second harmonic decoding module  520 , a first adder  530 , a CELP decoding module  540 , and a second adder  550 .  
         [0072]     The inverse quantizers  501 ,  502 ,  503 ,  504 ,  505 , and  506  can constitute an inverse quantization unit for inversely quantizing a plurality of parameters for restoring an audio signal.  
         [0073]     The first harmonic coding module  510  performs harmonic decoding using an LPC coefficient output from the LPC coefficient inverse quantizer  501  and a phase vector output from the phase index inverse quantizer  502  to output the restored audio signal including harmonics.  
         [0074]     To this end, the first harmonic coding module  510  includes an LPC coefficient/amplitude converter  511  and a harmonic synthesizer  512 .  
         [0075]     The LPC coefficient/amplitude converter  511  converts the LPC coefficient into a amplitude vector Â of a sine dictionary. The harmonic synthesizer  512  synthesizes the phase vector {circumflex over (φ)} output from the phase index inverse quantizer  502  with the amplitude vector Â of the sine dictionary output from the LPC/amplitude converter  511  using Equation 8 and outputs an audio signal including harmonics. The output audio signal including harmonics is output to the first adder  530 .  
         [0076]     The second harmonic coding module  520  performs harmonic coding based on the LPC coefficient output from the LPC coefficient inverse quantizer  501 , a harmonic index output from the harmonic index inverse quantizer  503 , and a first gain value output from the first gain index inverse quantizer  504 .  
         [0077]     To this end, the second harmonic coding module  520  includes a harmonic code generator  521 , a first multiplier  522 , and a first synthesis filter  523 .  
         [0078]     The harmonic code generator  521  includes a plurality of harmonic codes and generates a harmonic code based on the input harmonic index. The first multiplier  522  multiplies the generated harmonic code by the first gain value.  
         [0079]     The first synthesis filter  523  performs synthesis filtering on the signal output from the first multiplier  522  based on the input LPC coefficient and outputs the synthesized and filtered audio signal to the first adder  530 . If the audio signal output from the first multiplier  522  is s h (n), the LPC coefficient is a and the synthesized and filtered audio signal is s 1 (n), the synthesis filtering can be defined by Equation 9 
                   s   1     ⁡     (   n   )       =         s   h     ⁡     (   n   )       -       ∑     i   =   1     p     ⁢       a   i     ⁢       s   1     ⁡     (     n   -   i     )               ,           (   9   )             
 
 where p is the order of the LPC coefficient. 
 
         [0080]     The first adder  530  adds the signal output from the first harmonic coding module  510  to the signal output from the second harmonic coding module  520  and outputs the added result to the second adder  550 .  
         [0081]     The CELP decoding unit  540  performs CELP decoding based on the stochastic index output from the stochastic index inverse quantizer  505  and the second gain value output from the second gain index inverse quantizer  506 .  
         [0082]     To this end, the CELP decoding module  540  includes a stochastic codebook  541 , a multiplier  542 , and a second synthesis filter  543 .  
         [0083]     The stochastic codebook  541  includes a plurality of stochastic codebooks and outputs a stochastic codebook corresponding to the stochastic index.  
         [0084]     The second multiplier  542  multiplies the second gain value by the stochastic codebook. The second synthesis filter  543  provides the synthesized audio signal obtained by performing synthesis filtering on the signal output from the second multiplier  542  based on the LPC coefficient using Equation 9 to the second adder  550 .  
         [0085]     The second adder  550  adds the signal output from the first adder  530  to the signal output from the CELP decoding module  540  to restore the audio signal and outputs the restored audio signal.  
         [0086]      FIG. 6  is a flowchart illustrating an audio coding method according to another embodiment of the present invention. The audio coding method illustrated in  FIG. 6  will now be described with reference to  FIG. 1  for ease of explanation only.  
         [0087]     In operation  601 , a pitch of an input audio signal is analyzed to obtain a pitch lag.  
         [0088]     In operation  602 , it is determined whether harmonics are included in the input audio signal to classify the input audio signal and bits allocated to the first harmonic coding module  140 , the second harmonic coding module  160 , and the CELP module  180  based on the classification.  
         [0089]     In operation  603 , harmonic coding is performed with respect to the input audio signal by the first harmonic coding module  140  using the pitch lag obtained in operation  601 , without analyzing an LPC coefficient. That is, harmonic analysis with respect to the input audio signal is performed, the amplitude of the sine dictionary detected by harmonic analysis is converted into an LPC coefficient, the LPC coefficient is quantized and converted into the amplitude vector, and harmonic synthesis is performed. The quantized LPC coefficient is used in second harmonic coding and CELP coding.  
         [0090]     In operation  604 , a difference audio signal obtained as a difference between the input audio signal and the harmonic coding result obtained in operation  603  is set as a first difference audio signal, an LPC coefficient of the first difference audio signal is analyzed, and harmonic coding is performed on the first difference audio signal by the second harmonic coding module  160 . Here, the LPC coefficient of the first difference audio signal is extracted using the quantized LPC coefficient detected in operation  603 .  
         [0091]     In operation  605 , a difference audio signal obtained as a difference between the harmonic coding result obtained from the first difference audio signal and the input audio signal is set as a second difference audio signal, and the second difference audio signal is CELP coded by the CELP module  180 . In the CELP coding, the LPC coefficient of the second difference audio signal is extracted using the quantized LPC coefficient detected in operation  603 .  
         [0092]     In operation  606 , the plurality of parameters obtained in operations  603 ,  604 , and  605  are transmitted in order to decode an audio signal. The plurality of parameters include the quantized LPC coefficient index, a quantized phase index, a quantized harmonic index, a first quantized gain index, a quantized stochastic index, and a second quantized gain index.  
         [0093]      FIG. 7  is a flowchart illustrating an audio decoding method according to another embodiment of the present invention. The audio decoding method illustrated in  FIG. 7  will now be described with reference to  FIG. 5  for ease of explanation only.  
         [0094]     A plurality of parameters for restoring an audio signal are received in operation  701 , and each of the plurality of received parameters is inverse quantized in operation  702 .  
         [0095]     In operation  703 , harmonic decoding is performed by the first harmonic coding module  510  based on an LPC coefficient and a phase value obtained in operation  702 . In operation  704 , harmonic decoding is performed by the second harmonic coding module  520  based on the LPC coefficient, a harmonic index, and a first gain value obtained in operation  702 . In operation  705 , an audio signal in which the first harmonic decoding result obtained in operation  703  is added to the second harmonic decoding result obtained in operation  704  is obtained. In operation  706 , CELP decoding is performed by the CELP decoding module  540  based on a stochastic index and a second gain value obtained in operation  702 .  
         [0096]     In operation  707 , the addition result obtained in operation  705  is added to the CELP decoding result obtained in operation  706  to restore the audio signal.  
         [0097]     Embodiments of the present invention can also be embodied as computer readable code on a computer readable recording medium. The computer readable recording medium is any data storage device that can store data which can be thereafter read by a computer system. Examples of the computer readable recording medium include read-only memory (ROM), random-access memory (RAM), CD-ROMs, magnetic tapes, floppy disks, optical data storage devices, and carrier waves (such as data transmission through the Internet). The computer readable recording medium can also be distributed over network coupled computer systems so that the computer readable code is stored and executed in a distributed fashion. Also, functional programs, code, and code segments for accomplishing the present invention can be easily construed by programmers skilled in the art to which the present invention pertains.  
         [0098]     According to the above-described embodiments of the present invention, harmonic analysis is performed twice such that more harmonics can be searched for using the same bits.  
         [0099]     Allocation of bits used in harmonic coding is variably performed according to whether harmonics are included in the input audio signal such that a coarse granularity scalability function can be easily supported and harmonic sound quality can be optimised.  
         [0100]     In addition, after harmonic coding in which the LPC coefficient is not analysed is performed, harmonic coding in which the LPC coefficient is analysed is performed, and then, CELP coding is performed such that pitch halving prediction or pitch doubling prediction can be prevented and lowering of sound quality can be minimized.  
         [0101]     Although a few embodiments of the present invention have been shown and described, the present invention is not limited to the described embodiments. Instead, it would be appreciated by those skilled in the art that changes may be made to these embodiments without departing from the principles and spirit of the invention, the scope of which is defined by the claims and their equivalents.

Technology Category: 3