Abstract:
An audio signal processing device comprises a discontinuity detector configured to determine an occurrence of a discontinuity from a sudden increase of an amplitude of decoded audio obtained by decoding the first audio packet which is received correctly after an occurrence of a packet loss, and a discontinuity corrector for correcting the discontinuity of the decoded audio.

Description:
RELATED APPLICATIONS 
       [0001]    This application is a continuation of PCT/JP2014/077215 filed on Oct. 10, 2014, which claims priority to Japanese Application No. 2013-224120 filed on Oct. 29, 2013. The entire contents of these applications are incorporated herein by reference. 
     
    
     TECHNICAL FIELD 
       [0002]    The present invention relates to an audio signal processing device, an audio signal processing method, and an audio signal processing program for processing an audio signal. 
       BACKGROUND ART 
       [0003]    In transmission of a coded and packetized audio signal through an Internet network with an IP (Internet Protocol) phone, a packet can be lost because of a network congestion or the like (this phenomenon will be referred to hereinafter as “packet loss”). With an occurrence of a packet loss, necessary audio codes are lost resulting in a failure in decoding of audio, thereby causing an audio discontinuity. A technology for preventing an audio discontinuity caused by a packet loss is an audio packet loss concealment technology. The audio packet loss concealment technology is designed to detect a packet loss and generate a pseudo audio signal corresponding to the lost packet (which will be referred to hereinafter as “concealment signal”). 
         [0004]    When an audio encoding technique used is a technique of performing audio encoding while updating internal states of encoder/decoder, encoding parameters to be originally received are not obtained and thus the audio packet loss concealment technology includes performing an update of the internal states of the decoder by use of artificially-generated parameters as well. 
         [0005]    The CELP (Code Excited Linear Prediction) encoding is widely used as a technique for performing the audio encoding while updating the internal states of encoder/decoder. In the CELP encoding, an autoregressive model is assumed, and an excitation signal e(n) is filtered by an all-pole synthesis filter a(i) to synthesize an audio signal. Namely, the audio signal s(n) is synthesized according to the below equation. In the equation below, a(i) represents linear prediction coefficients (LP (Linear Prediction) coefficients) and the degree to be used is a value such as P=16. 
         [0000]    
       
         
           
             
               
                 
                   
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         [0006]    In the CELP encoding, the internal states stored include ISF (Immittance Spectral Frequency) parameters as mathematically equivalent representation of the linear prediction coefficients, and a past excitation signal. With an occurrence of a packet loss, these are artificially generated, and there arises a deviation from the original parameters that would be obtained by decoding. An inconsistency of a synthesized audio caused by a deviation of the parameters is perceived as a noise by a listener, which significantly degrades the subjective quality. 
         [0007]    The paragraphs below will describe a configuration and an operation of an audio decoder to perform the audio packet loss concealment, using an example where the CELP encoding is used as the audio encoding technique. 
         [0008]    A configuration diagram and an operation of the audio decoder are shown in  FIG. 1  and  FIG. 2 . As shown in  FIG. 1 , an audio decoder  1  has a packet loss detector  11 , an audio code decoder  12 , a concealment signal generator  13 , and an internal state buffer  14 . 
         [0009]    The packet loss detector  11 , when receiving an audio packet correctly, sends a control signal, and audio codes included in the audio packet, to the audio code decoder  12  (normal reception: YES in step S 100  in  FIG. 2 ). Thereafter, the audio code decoder  12  performs decoding of the audio codes and updating of the internal states as described below (steps S 200  and S 400  in  FIG. 2 ). On the other hand, the packet loss detector  11 , when failing to receive an audio packet correctly, sends a control signal to the concealment signal generator  13  (packet loss: NO in step S 100  in  FIG. 2 ). Thereafter, the concealment signal generator  13  generates a concealment signal and updates the internal states as described below (steps S 300  and S 400  in  FIG. 2 ). The processes of steps S 100  to S 400  in  FIG. 2  are repeated to the end of communication (or until step S 500  results in a determination of YES). 
         [0010]    The audio codes include at least encoded ISF parameters 
         [0000]      {dot over (ω)} i ,  [Mathematical Equation 2]
 
         [0000]    (Equation 2 is incomplete)
 
encoded pitch lags T j   p  of the first to fourth subframes, encoded adaptive codebook gains g j   p  of the first to fourth subframes, encoded fixed codebook gains g j   c  of the first to fourth subframes, and encoded fixed codebook vectors c j (n) of the first to fourth subframes. The ISF parameters may be replaced by LSF (line spectral frequency) parameters which are mathematically equivalent representation thereof. Although the discussion below uses the ISF parameters, the same discussion may also be true for the case using the LSF parameters.
 
         [0011]    The internal state buffer includes past ISF parameters 
         [0000]      {dot over (ω)} i   −1   [Mathematical Equation 3]
 
         [0000]    and, as equivalent representation of 
         [0000]      {dot over (ω)} i   − ,  [Mathematical Equation 4]
 
         [0012]    ISP (Immittance Spectral Pair) Parameters 
         [0000]      {dot over (q)} i   −1 ,  [Mathematical Equation 5]
 
         [0013]    ISF Residual Parameters 
         [0000]      {dot over (r)} i   −1 ,  [Mathematical Equation 6]
 
         [0000]    past pitch lags T j   p , past adaptive codebook gains g j   p , past fixed codebook gains g j   c , and an adaptive codebook u(n). It is determined, depending upon a design principle, how many subframes of the past parameters should be included. It is assumed in the present specification that one frame includes four subframes, but another value may be adopted depending upon the design principle. 
         [0014]    &lt;Case of Normal Reception&gt; 
         [0015]      FIG. 3  shows an exemplary functional configuration of the audio code decoder  12 . As shown in this  FIG. 3 , the audio code decoder  12  has an ISF decoder  120 , a stability processor  121 , an LP coefficient calculator  122 , an adaptive codebook calculator  123 , a fixed codebook decoder  124 , a gain decoder  125 , an excitation vector synthesizer  126 , a post-filter  127 , and a synthesis filter  128 . It should be noted, however, that the post-filter  127  is not an indispensable constitutive element. In  FIG. 3 , for convenience of explanation, the internal state buffer  14  is indicated by a double-dot line inside the audio code decoder  12 . However, the internal state buffer  14  is not included inside the audio code decoder  12 , but is indeed the internal state buffer  14  itself shown in  FIG. 1 . The same is also true in the configuration diagrams of the audio code decoder hereinafter. 
         [0016]    A configuration diagram of the LP coefficient calculator  122  is shown in  FIG. 4  and a processing flow of calculation of LP coefficients from the encoded ISF parameters is shown in  FIG. 5 . As shown in  FIG. 4 , the LP coefficient calculator  122  has an ISF-ISP converter  122 A, an ISP interpolator  122 B, and an ISP-LPC converter  122 C. 
         [0017]    First described are a functional configuration and its operation associated with the process of calculating the LP coefficients from the encoded ISF parameters ( FIG. 5 ). 
         [0018]    The ISF decoder  120  decodes the encoded ISF parameters to obtain the ISF residual parameters 
         [0000]      {dot over (r)} i   0   [Mathematical Equation 7]
 
         [0000]    and calculates the ISF parameters 
         [0000]      {dot over (ω)} i   [Mathematical Equation 8]
 
         [0000]    in accordance with the following equation (step S 1  in  FIG. 5 ). Here, mean i  represents mean vectors obtained in advance by learning or the like. 
         [0000]      {dot over (ω)} i =mean i   +{dot over (r)}   i   0 +⅓ {dot over (r)}   i   −1   [Mathematical Equation 9]
 
         [0019]    The example of using an MA prediction for the calculation of the ISF parameters is described herein, but it is also possible to adopt a configuration to perform calculation of the ISF parameters using an AR prediction as described below. Here, the ISF parameters of the immediately preceding frame are denoted by 
         [0000]      {dot over (ω)} i   −1   [Mathematical Equation 10]
 
         [0000]    and weight factors of the AR prediction by p i . 
         [0000]      {dot over (ω)} i =mean  i   p   i ({dot over (ω)} i   −1 −mean i )  [Mathematical Equation 11]
 
         [0020]    The stability processor  121  performs a process according to the below equation so as to place a distance of not less than 50 Hz between elements of the ISF parameters in order to secure stability of the filter (step S 2  in  FIG. 5 ). The ISF parameters are indicative of a line spectrum representing the shape of an audio spectrum envelope, and as the distance between them becomes shorter, peaks of the spectrum become larger, causing resonance. For this reason, the process for securing stability becomes necessary to prevent gains from becoming too large at the peaks of the spectrum. Here, min_dist represents a minimum ISF distance, and isf_min represents a minimum of ISF necessary for securing the distance of min_dist. isf_min is successively updated by adding the distance of min_dist to a value of neighboring ISF. On the other hand, isf_max represents a maximum of ISF necessary for securing the distance of min_dist. isf_max is successively updated by subtracting the distance of min_dist from a value of neighboring ISF. 
         [0000]      isf_min=min_dist=50 
         [0000]      for i=0 to 14 
         [0000]      if {dot over (ω)} i &lt;isf_min then {dot over (ω)} i =isf_min
 
         [0000]      isf_min={dot over (ω)} i +min_dist
 
         [0000]      isf_max=6400−min_dist
 
         [0000]      if {dot over (ω)} 14 &gt;isf_max
 
         [0000]      for i=14 down to 1 
         [0000]      if {dot over (ω)} i &gt;isf_max then {dot over (ω)} i =isf_max
 
         [0000]      isf_max={dot over (ω)} i −min_dist  [Mathematical Equation 12]
 
         [0021]    The ISF-ISP converter  122 A in the LP coefficient calculator  122  converts 
         [0000]      {dot over (ω)} i   [Mathematical Equation 13]
 
         [0000]    into ISP parameters 
         [0000]      {dot over (q)} i   [Mathematical Equation 14]
 
         [0000]    in accordance with the following equation (step S 3  in  FIG. 5 ). Here, C is a constant determined in advance. 
         [0000]      {dot over (q)} i =cos ( C ·{dot over (ω)} i )  [Mathematical Equation 15]
 
         [0022]    The ISP interpolator  122 B calculates the ISP parameters for the respective subframes from the past ISP parameters 
         [0000]      {dot over ( 1 )} i   −1   [Mathematical Equation 16]
 
         [0000]    included in the internal state buffer  14  and the foregoing ISP parameters 
         [0000]      {dot over (q)} i   [Mathematical Equation 17]
 
         [0000]    in accordance with the below equation (step S 4  in  FIG. 5 ). Other coefficients may be used for the interpolation. 
         [0000]        q   i   (1) =0.75 ·{dot over (q)}   i   −1 +0.25 ·{dot over (q)}   i    
         [0000]        q   i   (2) =0.5 ·{dot over (q)}   i   −1 +0.5 ·{dot over (q)}   i    
         [0000]        q   i   (3) =0.25 ·{dot over (q)}   i   −1 +0.75 ·{dot over (q)}   i    
         [0000]        q   i   (4)   ={dot over (q)}   i   [Mathematical Equation 18]
 
         [0023]    The ISP-LPC converter  122 C converts the ISP parameters for the respective subframes into LP coefficients 
         [0000]        {dot over (a)}   i   j (0 &lt;i≦P, 0 ≦j &lt;4)[Mathematical Equation 19] 
         [0000]    (step S 5  in  FIG. 5 ). A specific conversion procedure to be used can be the processing procedure described in Non Patent Literature 1. The number of subframes included in a look-ahead signal is assumed to be 4 herein, but the number of subframes may differ, depending upon the design principle. 
         [0024]    Next described are other configurations and operations in the audio code decoder  12 . 
         [0025]    The adaptive codebook calculator  123  decodes encoded pitch lags to calculate the pitch lags T j   P  of the first to fourth subframes. Then, the adaptive codebook calculator  123  uses the adaptive codebook u(n) to calculate adaptive codebook vectors for the respective subframes in accordance with the below equation. The adaptive codebook vectors are calculated by interpolating the adaptive codebook u(n) by a FIR filter Int(i). Here, the length of the adaptive codebook is denoted by N adapt . The filter Int(i) used for the interpolation is an FIR filter with a predetermined length 2l+1, and L′ presents the sample number of the subframes. By using the interpolation filter Int(i), the pitch lags can be utilized to the accuracy of decimal places. For the details of the interpolation filter, the method described in Non Patent Literature 1 can be referred to. 
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         [0026]    The fixed codebook decoder  124  decodes the encoded fixed codebook vectors to acquire the fixed codebook vectors c j (n) of the first to fourth subframes. 
         [0027]    The gain decoder  125  decodes the encoded adaptive codebook gains and the encoded fixed codebook gains to acquire the adaptive codebook gains and fixed codebook gains of the first to fourth subframes. For example, the decoding of the adaptive codebook gains and the fixed codebook gains can be carried out, for example, by the below technique described in Non Patent Literature 1. Since the below technique described in Non Patent Literature 1 does not use the interframe prediction as used in gain encoding of AMR-WB, it can enhance packet loss resistance. 
         [0028]    For example, the gain decoder  125  acquires the fixed codebook gain in accordance with the below processing flow. 
         [0029]    First, the gain decoder  125  calculates the power of the fixed codebook vector. Here, the length of the subframe is defined as N s . 
         [0000]    
       
         
           
             
               
                 
                   
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         [0030]    Next, the gain decoder  125  decodes the vector-quantized gain parameter to acquire the adaptive codebook gain 
         [0000]      ĝ p   [Mathematical Equation 22]
 
         [0000]    and the quantized fixed codebook gain 
         [0000]      Ê i .  [Mathematical Equation 23]
 
         [0031]    It then calculates a predictive fixed codebook gain as described below from the quantized fixed codebook gain and the aforementioned power of the fixed codebook vector. 
         [0000]        g′   c =10 0.05(Ê     i     −E     c   )  [Mathematical Equation 24]
 
         [0032]    Finally, the gain decoder  125  decodes the prediction coefficient 
         [0000]      ŷ  [Mathematical Equation 25]
 
         [0000]    and multiplies it to the prediction gain to acquire the fixed codebook gain. 
         [0000]        ĝ   c   =ŷ·g′   c   [Mathematical Equation 26]
 
         [0033]    The excitation vector synthesizer  126  multiplies the adaptive codebook vector by the adaptive codebook gain and multiplies the fixed codebook vector by the fixed codebook gain and calculates a sum of them to acquire an excitation signal, as expressed by the following equation. 
         [0000]        e   j ( n )= g   j   p   ·v   j ( n )+ g   j   c   ·c   j ( n )  [Mathematical Equation 27]
 
         [0034]    The post-filter  127  subjects the excitation signal vectors, for example, to post-processes such as processes of pitch enhancement, noise enhancement, and low-frequency enhancement. The pitch enhancement, the noise enhancement, and the low-frequency enhancement can be effected by use of the techniques described in Non Patent Literature 1. 
         [0035]    The synthesis filter  128  synthesizes a decoded signal with the excitation signal as a drive audio source, by linear prediction inverse filtering. 
         [0000]    
       
         
           
             
               
                 
                   
                     
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         [0036]    If a pre-emphasis is done in the encoder, a de-emphasis is carried out. 
         [0000]        ŝ   de-emph ( n )= ŝ ( n )+β· ŝ ( n −1)  [Mathematical Equation 29]
 
         [0037]    On the other hand, if a pre-emphasis is not done in the encoder, a de-emphasis is not carried out. 
         [0038]    The paragraphs below will describe the operation concerning an internal state update. 
         [0039]    In order to interpolate parameter upon an occurrence of packet loss, the LP coefficient calculator  122  updates the internal states of the ISF parameters by vectors calculated by the following equation. 
         [0000]    
       
         
           
             
                 
             
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         [0040]    Here, ω i   (−j)  represents the ISF parameters j frames prior, which are stored in the buffer. ω i   C  represents the ISF parameters in speech intervals obtained in advance by learning or the like. β is a constant and can be a value of, e.g., 0.75, to which the value is not necessarily limited. ω i   C  and β may be varied by an index to express a property of an encoding target frame, for example, as in the ISF concealment described in Non Patent Literature 1. 
         [0041]    Furthermore, the LP coefficient calculator  122  also updates the internal states of the ISF residual parameters in accordance with the following equation. 
         [0000]        {dot over (r)}   i   −1   =r   i   0   [Mathematical Equation 31]
 
         [0042]    The excitation vector synthesizer  126  updates the internal states by the excitation signal vectors in accordance with the below equation. 
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         [0043]    Furthermore, the excitation vector synthesizer  126  updates the internal states of the gain parameters by the following equation. 
         [0000]        g   c   (−M     in     +j)   =g   c   j   [Mathematical Equation 33]
 
         [0044]    The adaptive codebook calculator  123  updates the internal states of the parameters of the pitch lags by the following equation. 
         [0000]        T   p   (−M     la     +j)   =T   p   j   [Mathematical Equation 34]
 
         [0000]    The range of j is defined as (−2≦j &lt;M la ) but different values may be selected as the range of j, depending upon the design principle. 
         [0045]    &lt;Case of Packet Loss&gt; 
         [0046]      FIG. 6  shows an exemplary functional configuration of the concealment signal generator  13 . As shown in this  FIG. 6 , the concealment signal generator  13  has an LP coefficient interpolator  130 , a pitch lag interpolator  131 , a gain interpolator  132 , a noise signal generator  133 , a post-filter  134 , a synthesis filter  135 , an adaptive codebook calculator  136 , and an excitation vector synthesizer  137 . It should be noted, however, that the post-filter  134  is not an indispensable constitutive element. 
         [0047]    The LP coefficient interpolator  130  calculates 
         [0000]      {dot over (ω)} i   [Mathematical Equation 35]
 
         [0000]    by the following equation. In this respect, ω i   (−j)  represents the ISF parameters j frames prior, which are stored in the buffer. 
         [0000]      {dot over (ω)} i =αω i   (−1) +(1−α){right arrow over (ω)} i   [Mathematical Equation 36]
 
         [0048]    In this equation, 
         [0000]      {right arrow over (ω)} i   [Mathematical Equation 37]
 
         [0000]    represents the internal states of the ISF parameters calculated upon normal reception of a packet. α is also a constant and can be a value of, e.g., 0.9 to which the value is not necessarily limited. a may be varied by an index to express a property of an encoding target frame, for example, as in the ISF concealment described in Non Patent Literature 1. 
         [0049]    The procedure of obtaining the LP coefficients from the ISF parameters is the same as performed in the case of normal reception of a packet. 
         [0050]    The pitch lag interpolator  131  uses the internal state parameters about the pitch lags 
         [0000]        T   p   (−M     la     +j)   [Mathematical Equation 38]
 
         [0000]    to calculate predictive values of the pitch lags 
         [0000]      {circumflex over (T)} p .  [Mathematical Equation 39]
 
         [0051]    A specific processing procedure to be used can be the technique disclosed in Non Patent Literature 1. 
         [0052]    In order to interpolate the fixed codebook gains, the gain interpolator  132  can use the technique according to the below equation as described in Non Patent Literature 1. 
         [0000]        g   s =0.4 ·g   c   −1 +0.3 ·g   c   −2 +0.2 ·g   c   −3 +0.1 ·g   c   −4   [Mathematical Equation 40]
 
         [0053]    The noise signal generator  133  generates white noise for the same length as the fixed codebook vectors and uses the resultant noise for the fixed codebook vectors. 
         [0054]    The operations of the post-filter  134 , the synthesis filter  135 , the adaptive codebook calculator  136 , and the excitation vector synthesizer  137  are the same as those in the aforementioned case of normal reception of a packet. 
         [0055]    The internal state update is the same as performed in the case of normal reception of a packet, except for an update of the ISF residual parameters. The updating of the ISF parameters is carried out in accordance with the following equation by the LP coefficient interpolator  130 . 
         [0000]        {dot over (r)}   i   0 ={dot over (ω)} i   0 −mean i −⅓ {dot over (r)}   i   −1   [Mathematical Equation 41]
 
       CITATION LIST 
     Patent Literatures 
       [0056]    Patent Literature 1: International Publication WO 2002/035520 
         [0057]    Patent Literature 2: International Publication WO 2008/108080 
       Non Patent Literature 
       [0058]    Non Patent Literature 1: ITU-T Recommendation G.718, June 2008 
       SUMMARY OF INVENTION 
     Technical Problem 
       [0059]    As described above, since the CELP encoding involves the internal states, a degradation of audio quality occurs because of a deviation between the parameters obtained by interpolations implemented upon a packet loss and the parameters that would have been used for decoding. Particularly, as to the ISF parameters, intraframe/interframe predictive encoding is carried out, and thus there is the problem that an influence by a packet loss continues even after recovery from the packet loss. 
         [0060]    More specifically, a problem of a sudden increase of power is identified in the first frame after recovery from a packet loss occurring in the vicinity of an audio start portion. This is caused for the following reason: That is, in the audio start portion where the power of the excitation signal becomes high, the impulse response of the LP coefficients calculated from the ISF coefficients obtained by the interpolation process upon a packet loss has a higher gain than the one that would have been originally expected for the decoder. This is perceived, according to the subjective quality standard, as an unpleasant discontinuity of audio. 
         [0061]    The method described in Patent Literature 1 generates the interpolated ISF coefficients for a lost frame. However, since the ISF parameters are generated by a normal decoding process for the first frame after recovery from the loss, it fails to suppress the sudden increase of power. 
         [0062]    On the other hand, the method described in Patent Literature 2 transmits a gain adjustment parameter (normalized prediction residual power) obtained on the encoding side and uses it for a power adjustment on the decoding side, thereby controlling the power of the excitation signal of a lost packet frame and enabling prevention of the sudden increase of power. 
         [0063]      FIG. 7  shows an exemplary functional configuration of an audio decoder  1 X implemented by the technology of Patent Literature 2, and  FIG. 8  shows an exemplary functional configuration of a concealment signal generator  13 X. In Patent Literature 2, an audio packet includes auxiliary information of at least a normalized prediction residual power in addition to the parameters described in the conventional technique. 
         [0064]    A normalized prediction residual power decoder  15  provided in the audio signal generator  1 X decodes the auxiliary information of the normalized prediction residual power from a received audio packet to calculate a reference normalized prediction residual power, and outputs it to the concealment signal generator  13 X. 
         [0065]    Since the constitutive elements of the concealment signal generator  13 X, other than normalized prediction residual adjuster  138 , are the same as those in the aforementioned conventional technology, only the normalized prediction residual adjuster  138  will be described below. 
         [0066]    The normalized prediction residual adjuster  138  calculates the normalized prediction residual power from the LP coefficients output by the LP coefficient interpolator  130 . Next, the normalized prediction residual adjuster  138  calculates a synthesis filter gain adjustment coefficient, using the normalized prediction residual power and the reference normalized prediction residual power. Finally, the normalized prediction residual adjuster  138  multiplies the excitation signal by the synthesis filter gain adjustment coefficient and output the result to the synthesis filter  135 . 
         [0067]    The above-described technology of Patent Literature 2 can control the power of the concealment signal upon an occurrence of a packet loss in the same manner as performed in the normal reception. However, it is difficult to secure a bit rate necessary for transmission of the foregoing gain adjustment parameter in the process of low-bit-rate audio encoding. In addition, since it is the processing in the concealment signal generator, it is difficult to deal with a sudden change of power caused by a disagreement of the ISF parameters in a recovery frame. 
         [0068]    An object of the present invention is therefore to reduce a discontinuity of audio which can occur upon recovery from a packet loss at the audio start point, and thereby improve the subjective quality. 
       Solution to Problem 
       [0069]    An audio signal processing device according to one embodiment of the present invention comprises: a discontinuity detector configured to determine an occurrence of a discontinuity occurring with a sudden increase of the amplitude of a decoded audio obtained by decoding a first audio packet which is received correctly after an occurrence of a packet loss; and a discontinuity corrector configured to correct the discontinuity of the decoded audio. 
         [0070]    The discontinuity detector may determine an occurrence of a discontinuity of the decoded audio with the power of an excitation signal. 
         [0071]    The discontinuity detector may detect an occurrence of a discontinuity of the decoded audio with quantized codebook gains used for calculation of an excitation signal. 
         [0072]    The audio signal processing device may further comprise: an auxiliary information decoder configured to decode auxiliary information for determination on an occurrence of a discontinuity transmitted from an encoder, and the discontinuity detector may determine an occurrence of a discontinuity of the decoded audio, using the auxiliary information decoded and output as an auxiliary information code by the auxiliary information decoder. 
         [0073]    The discontinuity corrector may correct ISF parameters or LSF parameters (hereinafter referred to as “ISF/LSF parameters”) according to a result of determination on an occurrence of a discontinuity. 
         [0074]    More specifically, the discontinuity corrector may change a distance between elements of the ISF/LSF parameters given for ensuring stability of a synthesis filter, according to a result of determination on an occurrence of a discontinuity. 
         [0075]    At this time, the discontinuity corrector may extend the distance between the elements of the ISF/LSF parameters given for ensuring the stability of the synthesis filter to become larger than an ordinary distance given for ensuring stability. 
         [0076]    For the distance between the elements of the ISF/LSF parameters given for ensuring the stability of the synthesis filter, the discontinuity corrector may use a distance, which is obtained by equally dividing the ISF/LSF parameters into those of a predetermined length. 
         [0077]    Furthermore, the discontinuity corrector may replace a part of or all of the ISF/LSF parameters with predetermined vectors. 
         [0078]    An audio signal processing device according to one embodiment of the present invention comprises: an ISF/LSF quantizer configured to quantize ISF/LSF parameters; an ISF/LSF concealer configured to generate concealment ISF/LSF parameters, which are concealment information for the ISF/LSF parameters; a discontinuity detector configured to determine an occurrence of a discontinuity occurring in a first audio packet which is received correctly after an occurrence of a packet loss, using distances between the quantized ISF/LSF parameters obtained in the quantization process by the ISF/LSF quantizer and the concealment ISF/LSF parameters generated by the ISF/LSF concealer; and an auxiliary information encoder configured to encode auxiliary information for determination on an occurrence of a discontinuity. 
         [0079]    An audio signal processing device according to one embodiment of the present invention comprises: a discontinuity detector configured to determine an occurrence of a discontinuity occurring in a first audio packet which is received correctly after an occurrence of a packet loss; an auxiliary information encoder configured to encode auxiliary information for determination on an occurrence of a discontinuity; and an ISF/LSF quantizer configured to use past quantized ISF/LSF residual parameters for ISF/LSF quantization in a given frame when the discontinuity detector does not determine an occurrence of a discontinuity, and avoid using the past quantized ISF/LSF residual parameters for ISF/LSF quantization in the given frame when the discontinuity detector determines an occurrence of a discontinuity. 
         [0080]    An audio signal processing device according to one embodiment of the present invention comprises: an auxiliary information decoder configured to decode and output auxiliary information for determination on an occurrence of a discontinuity occurring in a first audio packet which is received correctly after an occurrence of a packet loss; a discontinuity corrector configured to correct the discontinuity of a decoded audio; and an ISF/LSF decoder configured to use past quantized ISF/LSF residual parameters for ISF/LSF calculation in a pertinent frame when the auxiliary information from the auxiliary information decoder does not indicate an occurrence of a discontinuity, and avoid using the past quantized ISF/LSF residual parameters for the ISF/LSF calculation in the pertinent frame when the auxiliary information from the auxiliary information decoder indicates an occurrence of a discontinuity. 
         [0081]    The audio signal processing device may adopt a configuration in which the audio signal processing device further comprises: a reception state determiner configured to determine packet reception states of a predetermined number of past frames; the discontinuity corrector corrects a discontinuity on the basis of a determination result of the packet reception states as well, in addition to a result of determination on an occurrence of a discontinuity. 
         [0082]    Now, the audio signal processing device according to one embodiment of the present invention may be taken as an invention associated with an audio signal processing method, and as an invention associated with an audio signal processing program, and can be described as below. 
         [0083]    An audio signal processing method according to one embodiment of the present invention is an audio signal processing method to be executed by an audio signal processing device, comprising: a step of determining an occurrence of a discontinuity of decoded audio occurring with a sudden increase of the amplitude of a decoded audio obtained by decoding a first audio packet which is received correctly after an occurrence of a packet loss; and a step of correcting the discontinuity of the decoded audio. 
         [0084]    An audio signal processing method according to one embodiment of the present invention is an audio signal processing method to be executed by an audio signal processing device, comprising: a step of quantizing ISF/LSF parameters; a step of generating concealment ISF/LSF parameters which are concealment information for the ISF/LSF parameters; a step of determining an occurrence of a discontinuity occurring in a first audio packet which is received correctly after an occurrence of a packet loss, using distances between quantized ISF/LSF parameters obtained in a quantization process of the ISF/LSF quantizer and the generated concealment ISF/LSF parameters; and a step of encoding auxiliary information for determination on an occurrence of a discontinuity. 
         [0085]    An audio signal processing method according to one embodiment of the present invention is an audio signal processing method to be executed by an audio signal processing device, comprising: a step of determining an occurrence of a discontinuity occurring in a first audio packet which is received correctly after an occurrence of a packet loss; a step of encoding auxiliary information for determination on an occurrence of a discontinuity; and a step of using past quantized ISF/LSF residual parameters for ISF/LSF quantization in a given frame when an occurrence of a discontinuity is not determined, and avoiding using the past quantized ISF/LSF residual parameters for the ISF/LSF quantization in the pertinent frame when an occurrence of a discontinuity is determined. 
         [0086]    An audio signal processing method according to one embodiment of the present invention is an audio signal processing method to be executed by an audio signal processing device, comprising: a step of decoding and outputting auxiliary information for determination on an occurrence of a discontinuity of decoded audio occurring in a first audio packet which is received correctly after an occurrence of a packet loss; a step of correcting the discontinuity of decoded audio; and a step of using past quantized ISF/LSF residual parameters for ISF/LSF calculation in a given frame when the auxiliary information does not indicate an occurrence of a discontinuity, and avoiding using the past quantized ISF/LSF residual parameters for the ISF/LSF calculation in the given frame when the auxiliary information indicates an occurrence of a discontinuity. 
         [0087]    An audio signal processing program according to one embodiment of the present invention is an audio signal processing program that programs a computer to operate as: a discontinuity detector operable to determine an occurrence of a discontinuity of decoded audio occurring with a sudden increase of amplitude of a decoded audio obtained by decoding a first audio packet which is received correctly after an occurrence of a packet loss; and a discontinuity corrector operable to correct the discontinuity of the decoded audio. 
         [0088]    An audio signal processing program according to one embodiment of the present invention is an audio signal processing program that programs a computer to operate as: an ISF/LSF quantizer operable to quantize ISF/LSF parameters; an ISF/LSF concealer operable to generate concealment ISF/LSF parameters which are concealment information for the ISF/LSF parameters; a discontinuity detector operable to determine an occurrence of a discontinuity occurring in a first audio packet which is received correctly after an occurrence of a packet loss, using distances between quantized ISF/LSF parameters obtained in a quantization process of the ISF/LSF quantizer and the concealment ISF/LSF parameters generated by the ISF/LSF concealer; and an auxiliary information encoder operable to encode auxiliary information for determination on an occurrence of a discontinuity. 
         [0089]    An audio signal processing program according to one embodiment of the present invention is an audio signal processing program that programs a computer to operate as: a discontinuity detector operable to determine an occurrence of a discontinuity occurring in a first audio packet which is received correctly after an occurrence of a packet loss; an auxiliary information encoder operable to encode auxiliary information for determination on an occurrence of a discontinuity; and an ISF/LSF quantizer operable to use past quantized ISF/LSF residual parameters for ISF/LSF quantization in a pertinent frame when the discontinuity detector does not determine an occurrence of a discontinuity, and avoid using the past quantized ISF/LSF residual parameters for the ISF/LSF quantization in the pertinent frame when the discontinuity detector determines an occurrence of a discontinuity. 
         [0090]    An audio signal processing program according to one embodiment of the present invention is an audio signal processing program that programs a computer to operate as: an auxiliary information decoder operable to decode and output auxiliary information for determination on an occurrence of a discontinuity of decoded audio occurring in a first audio packet which is received correctly after an occurrence of a packet loss; a discontinuity corrector operable to correct the discontinuity of decoded audio; and an ISF/LSF decoder operable to use past quantized ISF/LSF residual parameters for ISF/LSF calculation in a given frame when the auxiliary information from the auxiliary information decoder does not indicate an occurrence of a discontinuity, and avoid using the past quantized ISF/LSF residual parameters for the ISF/LSF calculation in the pertinent frame when the auxiliary information from the auxiliary information decoder indicates an occurrence of a discontinuity. 
       Advantageous Effect of Invention 
       [0091]    The present invention as described above can reduce a discontinuity of audio possibly occurring subsequent to recovery from a packet loss at the audio start point and thus improve the subjective quality. 
     
    
     
       BRIEF DESCRIPTION OF THE DRAWINGS 
         [0092]      FIG. 1  is a configuration diagram of the audio decoder. 
           [0093]      FIG. 2  is a processing flow of the audio decoder. 
           [0094]      FIG. 3  is a functional configuration diagram of the audio code decoder. 
           [0095]      FIG. 4  is a functional configuration diagram of the LP coefficient calculator. 
           [0096]      FIG. 5  is a processing flow of calculating the LP coefficients. 
           [0097]      FIG. 6  is a functional configuration diagram of the concealment signal generator. 
           [0098]      FIG. 7  is a configuration diagram of the audio decoder of Patent Literature 2. 
           [0099]      FIG. 8  is a functional configuration diagram of the concealment signal generator of Patent Literature 2. 
           [0100]      FIG. 9  is a functional configuration diagram of the audio code decoder in a first embodiment. 
           [0101]      FIG. 10  is a processing flow of the LP coefficient calculator in the first embodiment. 
           [0102]      FIG. 11  is a functional configuration diagram of the audio code decoder in the first embodiment. 
           [0103]      FIG. 12  is a processing flow of a second stability processor in modification example 1 of the first embodiment. 
           [0104]      FIG. 13  is a functional configuration diagram of the audio code decoder in a second embodiment. 
           [0105]      FIG. 14  is a functional configuration diagram of the LP coefficient calculator in the second embodiment. 
           [0106]      FIG. 15  is a processing flow of calculation of the LP coefficients in the second embodiment. 
           [0107]      FIG. 16  is a configuration diagram of an audio encoder in fourth embodiment. 
           [0108]      FIG. 17  is a configuration diagram of the audio encoder in the fourth embodiment. 
           [0109]      FIG. 18  is a configuration diagram of an LP analyzer/encoder in the fourth embodiment. 
           [0110]      FIG. 19  is a processing flow of the LP analyzer/encoder in the fourth embodiment. 
           [0111]      FIG. 20  is a functional configuration diagram of the audio code decoder in the fourth embodiment. 
           [0112]      FIG. 21  is a processing flow of the LP coefficient calculator in the fourth embodiment. 
           [0113]      FIG. 22  is a configuration diagram of the LP analyzer/encoder in the fifth embodiment. 
           [0114]      FIG. 23  is a processing flow of the LP analyzer/encoder in the fifth embodiment. 
           [0115]      FIG. 24  is a functional configuration diagram of the audio code decoder in the fourth embodiment. 
           [0116]      FIG. 25  is a processing flow of the LP coefficient calculator in the fifth embodiment. 
           [0117]      FIG. 26  is a configuration diagram of the audio decoder in the seventh embodiment. 
           [0118]      FIG. 27  is a processing flow of the audio decoder in the seventh embodiment. 
           [0119]      FIG. 28  is a functional configuration diagram of the audio code decoder in the seventh embodiment. 
           [0120]      FIG. 29  is a processing flow of calculation of the LP coefficients in the seventh embodiment. 
           [0121]      FIG. 30  is a drawing showing a hardware configuration example of a computer. 
           [0122]      FIG. 31  is an appearance diagram of the computer. 
           [0123]      FIGS. 32 ( a ), ( b ), ( c ), and ( d )  are drawings showing various examples of audio signal processing programs. 
       
    
    
     DETAILED DESCRIPTION OF THE PREFERRED EMBODIMENTS 
       [0124]    Preferred embodiments of an audio signal processing device, an audio signal processing method, and an audio signal processing program according to the present invention will be described below in detail using the drawings. The same elements will be denoted by similar reference signs in the description of the drawings to avoid duplicate descriptions. 
       First Embodiment 
       [0125]    The audio signal processing device in the first embodiment has the same configuration as the aforementioned audio decoder  1  shown in  FIG. 1  and has a novel feature in the audio code decoder, and thus the audio code decoder will be described below. 
         [0126]      FIG. 9  is a diagram showing a functional configuration of an audio code decoder  12 A in the first embodiment, and  FIG. 10  shows a flowchart of the LP coefficient calculation process. The audio code decoder  12 A shown in  FIG. 9  is configured by adding a discontinuity detector  129  to the aforementioned configuration of  FIG. 3 . Since the present embodiment differs from the conventional technology only in the LP coefficient calculation process, the operations of respective parts associated with the LP coefficient calculation process will be described below. 
         [0127]    A discontinuity detector  129  refers to a fixed codebook gain g c   0  acquired by decoding and a fixed codebook gain g c   −1  included in the internal states and compares a change of the gain with a threshold in accordance with the following equation (step S 11  in  FIG. 10 ). 
         [0000]      log ( g   c   0 ) −log ( g   c   −1 )&gt;  [Mathematical Equation 42]
 
         [0128]    When the gain change exceeds the threshold, the detector detects an occurrence of a discontinuity (also referred to hereinafter simply as “detects a discontinuity”) and outputs a control signal indicating a detection result of a discontinuity occurrence to the stability processor  121 . 
         [0129]    The following equation may be used for the comparison between the gain change and the threshold. 
         [0000]        g   c   0   −g   c   −1 &gt;Thres[Mathematical Equation 43] 
         [0130]    Furthermore, the comparison between the gain change and the threshold may be made by the following equation, where a g c   (c)  represents the maximum among the fixed codebook gains of the first to fourth subframes included in the current frame and a g c   (p)  represents the minimum among the fixed codebook gains included in the internal states. 
         [0000]      log ( g   c   (c) )−log ( g   c   (p) )&gt;Thres  [Mathematical Equation 44]
 
         [0131]    The flowing equation can also be used. 
         [0000]        g   c   (c)   −g   c   (p) &gt;Thres  [Mathematical Equation 45]
 
         [0132]    The above example of the first embodiment shows an example in which a discontinuity detection is conducted using the fixed codebook gain g c   −1  of the fourth subframe of the immediately preceding frame (lost frame) and the fixed codebook gain g c   0  of the first subframe of the current frame. However, comparison between the gain change and the threshold may be made using averages calculated from the fixed codebook gains included in the internal states and the fixed codebook gains included in the current frame. 
         [0133]    The ISF decoder  120  performs the same operation as in the conventional technology (step S 12  in  FIG. 10 ). 
         [0134]    The stability processor  121  corrects the ISF parameters by the following process when the discontinuity detector  129  detects a discontinuity (step S 13  in  FIG. 10 ). 
         [0135]    First, the stability processor  121  subjects the ISF parameters 
         [0000]      {dot over (ω)} i   −   [Mathematical Equation 46]
 
         [0000]    stored in the internal state buffer  14  to a process of expanding a distance between two adjacent element to become M −1  times wider than the ordinary distance. The process of placing a very wide distance than the ordinary distance provides an effect to suppress excessive peaks and dips in the spectrum envelope. Here, min_dist represents the minimum ISF distance, and isf_min represents the minimum of ISF necessary for securing the distance of min_dist. isf_min is successively updated by adding the distance of min_dist to a value of neighboring ISF. On the other hand, isf_max is the maximum of ISF necessary for securing the distance of min_dist. isf_max is successively updated by subtracting the distance of min_dist from a value of neighboring ISF. 
         [0000]      isf_min=min_dist=50 M   −1    
         [0000]      for i=0 to 14 
         [0000]      if {dot over (ω)} i   −1 &lt;isf_min then {dot over (ω)} i   −1 =isf_min
 
         [0000]      isf_min={dot over (ω)} i   −1 +min_dist
 
         [0000]      isf_max=6400−min_dist
 
         [0000]      if {dot over (ω)} 14   −1 &gt;isf_max
 
         [0000]      for i=14 down to 1 
         [0000]      if {dot over (ω)} i   −1 &gt;isf_max then {dot over (ω)} i   −1 =isf —max  
 
         [0000]      isf_max={dot over (ω)} i   −1 −min_dist  [Mathematical Equation 47 ]
 
         [0136]    Next, a stability processor  121  subjects the ISF parameters of the current frame to a process of expanding a distance between two adjacent element to become M 0  times wider than the ordinary distance. 1&lt;M 0 &lt;M −1  is assumed herein, but it is also possible to set one of r M −1  and M 0 to  1 and the other to a value larger than 1. 
         [0000]      isf_min=min_dist=50 M   0    
         [0000]      for i=0 to 14 
         [0000]      if {dot over (ω)} i   0 &lt;isf_min then {dot over (ω)} i   0 =isf_min
 
         [0000]      isf_min={dot over (ω)} i   0 +min_dist
 
         [0000]      isf_max=6400−min_dist
 
         [0000]      if {dot over (ω)} 14   0 &gt;isf_max
 
         [0000]      for i=14 down to 1 
         [0000]      if {dot over (ω)} i   0 &gt;isf_max then {dot over (ω)} i   0 =isf_max
 
         [0000]      isf_max={dot over (ω)} i   0 −min_dist  [Mathematical Equation 48]
 
         [0137]    Furthermore, the stability processor  121  performs the following process in the same manner as carried out in the ordinary decoding process, when the discontinuity detector detects no discontinuity. 
         [0000]      isf_min=min_dist=50 
         [0000]      for i=0 to 14 
         [0000]      if {dot over (ω)} i   0 &lt;isf_min then {dot over (ω)} i   0 =isf_min
 
         [0000]      isf_min={dot over (ω)} i   0 +min_dist
 
         [0000]      isf_max=6400−min_dist
 
         [0000]      if {dot over (ω)} 14   0 &gt;isf_max
 
         [0000]      for i=14 down to 1 
         [0000]      if {dot over (ω)} i   0 &gt;isf_max then {dot over (ω)} i =isf_max
 
         [0000]      isf_max={dot over (ω)} i   0 −min_dist  [Mathematical Equation 49]
 
         [0138]    The minimum distance placed between elements when a discontinuity is detected may be varied depending upon the frequency of ISF. The minimum distance placed between elements when a discontinuity is detected needs only to be different from the minimum distance placed between elements in the ordinary decoding process. 
         [0139]    The ISF-ISP converter  122 A in the LP coefficient calculator  122  converts the ISF parameters 
         [0000]      {dot over (ω)} i , {dot over (ω)} i   −1   [Mathematical Equation 50]
 
         [0000]    into the ISP parameters 
         [0000]      {dot over (q)} i ,{dot over (q)} i   −1 ,  [Mathematical Equation 51]
 
         [0000]    respectively, in accordance with the following equation (step S 14  in  FIG. 10 ). Here, C is a constant determined in advance. 
         [0000]        {dot over (q)}   i =cos ( C ·{dot over (ω)} i )  [Mathematical Equation 52]
 
         [0140]    The ISP interpolator  122 B calculates the ISP parameters for the respective subframes from the past ISP parameters 
         [0000]      {dot over (q)} i   −1   [Mathematical Equation 53]
 
         [0000]    and the foregoing ISP parameters 
         [0000]      {dot over (q)} i   [Mathematical Equation 54]
 
         [0000]    in accordance with the following equation (step S 15  in  FIG. 10 ). Other coefficients may be used for the interpolation. 
         [0000]        q   i   (1) =0.75 ·{dot over (q)}   i   −1 +0.25 ·{dot over (q)}   i    
         [0000]        q   i   (2) =0.5· {dot over (q)}   i   −1 +0.5 ·{dot over (q)}   i  
 
         [0000]        q   i   (3) =0.25 ·{dot over (q)}   i   −1 +0.75 ·{dot over (q)}   i    
         [0000]        q   i   (4)   ={dot over (q)}   i   [Mathematical Equation 55]
 
         [0141]    The ISP-LPC converter  122 C converts the ISP parameters for the respective subframes into the LP coefficients 
         [0000]      {dot over (α)} i   j (0 &lt;i≦P ,0 ≦j &lt;4)  [Mathematical Equation 56]
 
         [0000]    (step S 16  in  FIG. 10 ). Here, the number of subframes included in a look-ahead signal was assumed to be  4 , but the number of subframes may differ depending upon the design principle. A specific conversion procedure to be used can be the processing procedure described in Non Patent Literature 1. 
         [0142]    Furthermore, the ISF-ISP converter  122 A updates the ISF parameters stored in the internal state buffer  14   
         [0000]      {dot over (ω)} i   −1   [Mathematical Equation 57]
 
         [0000]    in accordance with the following equation. 
         [0000]      {dot over (ω)} i   −1 ={dot over (ω)} i   0   [Mathematical Equation 58]
 
         [0143]    At this time, even when a discontinuity is detected, the ISF-ISP converter  122 A may carry out the below procedure to update the ISF parameters 
         [0000]      {dot over (ω)} i   −1   [Mathematical Equation 59]
 
         [0000]    stored in the internal state buffer, using the calculation result of the ISF parameters. 
         [0000]      isf_min=min_dist=50 
         [0000]      for i=0 to 14 
         [0000]      if {dot over (ω)} i   0 &lt;isf_min then {dot over (ω)} i   0 =isf_min
 
         [0000]      isf_min={dot over (ω)} i   0 +min_dist
 
         [0000]      isf_max=6400−min_dist
 
         [0000]      if {dot over (ω)} 14   0 &gt;isf_max
 
         [0000]      for i=14 down to 1 
         [0000]      if {dot over (ω)} i   0 &gt;isf_max then {dot over (ω)} i   0 =isf_max
 
         [0000]      isf_max={dot over (ω)} i   0 −min_dist  [Mathematical Equation 60]
 
         [0144]    As in the above first embodiment, a discontinuity of decoded audio can be determined with the quantized codebook gains used in the calculation of the excitation signal and the ISF/LSF parameters (e.g., the distance between elements of the ISF/LSF parameters given for ensuring stability of the synthesis filter) can be corrected according to a result of the determination for a discontinuity. This reduces the discontinuity of audio which can occur upon recovery from a packet loss at the audio start point, and thereby improves the subjective quality. 
       Modification Example of First Embodiment 
       [0145]      FIG. 11  is a diagram showing a functional configuration of an audio code decoder  12 S according to a modification example of the first embodiment. Since it differs from the configuration of the conventional technology shown in  FIG. 3  only in the discontinuity detector  129  and the second stability processor  121 S, the operations of these will be described. The second stability processor  121 S has a gain adjustor  121 X and a gain multiplier  121 Y, and a processing flow of the second stability processor  121 S is shown in  FIG. 12 . 
         [0146]    The discontinuity detector  129  refers to the fixed codebook gain g c   0  obtained by decoding and the fixed codebook gain g c   −1  included in the internal states and compares the gain change with a threshold, in the same manner as performed by the discontinuity detector  129  in the first embodiment. Then, the discontinuity detector  129  sends to the gain adjustor  121 X, a control signal including information about whether the gain change exceeds the threshold. 
         [0147]    The gain adjustor  121 X reads from the control signal the information about whether the gain change exceeds the threshold, and, when the gain change exceeds the threshold, it outputs a predetermined gain g on  to the gain multiplier  121 Y. On the other hand, when the gain change does not exceed the threshold, the gain adjustor  121 X outputs a predetermined gain g off  to the gain multiplier  121 Y. This operation of the gain adjustor  121 X corresponds to step S 18  in  FIG. 12 . 
         [0148]    The gain multiplier  121 Y multiplies the synthesized signal output from the synthesis filter  128  by the foregoing gain g on  or gain g off  (step S 19  in  FIG. 12 ) and outputs the resultant decoded signal. 
         [0149]    Here, the audio code decoder may be configured such that the LP coefficient calculator  122  outputs the LP coefficients or the ISF parameters to feed them to the second stability processor  121 S (as indicated by a dotted line from the LP coefficient calculator  122  to the gain adjustor  121 X in  FIG. 11 ). In this case, the gains to be multiplied are determined using the LP coefficients or the ISF parameters calculated by the LP coefficient calculator  122 . 
         [0150]    By adding the second stability processor  121 S to the audio code decoder  12 S and adjusting the gain, depending upon whether the gain change exceeds the threshold as described in the above modification example, an appropriate decoded signal can be obtained. 
         [0151]    The second stability processor  121 S may be configured to multiply the excitation signal by the foregoing calculated gain and output the result to the synthesis filter  128 . 
       Second Embodiment 
       [0152]    An audio signal processing device according to the second embodiment has the same configuration as that of the aforementioned audio decoder  1  in  FIG. 1  and has a novel feature in an audio code decoder, and thus the audio code decoder will be described below.  FIG. 13  shows an exemplary functional configuration of the audio code decoder  12 B,  FIG. 14  shows an exemplary functional configuration associated with the calculation process of the LP coefficients, and  FIG. 15  shows a flow of the calculation process of the LP coefficients. The audio code decoder  12 B in  FIG. 13  is configured by adding the discontinuity detector  129  to the aforementioned configuration shown in  FIG. 3 . 
         [0153]    The ISF decoder  120  calculates the ISF parameters in the same manner as performed in the conventional technology (step S 21  in  FIG. 15 ). 
         [0154]    The stability processor  121  performs the process of placing a distance of not less than 50 Hz between elements of the ISF parameters 
         [0000]      {dot over (ω)} i   [Mathematical Equation 61]
 
         [0000]    in order to secure the stability of the filter in the same manner as performed in the conventional technology (step S 22  in  FIG. 15 ). 
         [0155]    The ISF-ISP converter  122 A converts the ISF parameters output by the stability processor  121  into the ISP parameters in the same manner as performed in the first embodiment (step S 23  in  FIG. 15 ). 
         [0156]    The ISP interpolator  122 B, in the same manner as performed in the first embodiment (step S 24  in  FIG. 15 ), calculates the ISP parameters for the respective subframes from the past ISP parameters 
         [0000]      {dot over (q)} i   −1   [Mathematical Equation 62]
 
         [0000]    and the ISP parameters 
         [0000]      {dot over (q)} i   [Mathematical Equation 63]
 
         [0000]    obtained by the conversion by the ISF-ISP converter  122 A. 
         [0157]    The ISP-LPC converter  122 C, in the same manner as performed in the first embodiment (step S 25  in  FIG. 15 ), converts the ISP parameters for the respective subframes into the LP coefficients 
         [0000]      {dot over (α)} i   j (0 &lt;i≦P, 0 ≦j &lt;4)  [Mathematical Equation 64]
 
         [0000]    Here, the number of subframes included in the look-ahead signal is assumed to be 4, but the number of subframes may differ depending upon the design principle. 
         [0158]    The internal state buffer  14  updates the ISF parameters stored in the past with the new ISF parameters. 
         [0159]    The discontinuity detector  129  reads the LP coefficients of the fourth subframe in the lost packet frame from the internal state buffer  14  and calculates the power of the impulse response of the LP coefficients of the fourth subframe in the lost packet frame. The LP coefficients of the fourth subframe in the lost packet frame to be used can be the coefficients output by the LP coefficient interpolator  130  included in the concealment signal generator  13  shown in  FIG. 6  and accumulated in the internal state buffer  14  upon the packet loss. 
         [0000]        E   −1 10 log (Σ n=0   L′−1   h   −1   2 ( n ))
 
         [0000]        h   −1 ( n )=δ( n )−Σ i=1   P {dot over (a)} i   (−1)   ·h   −1 ( n−i )  [Mathematical Equation 65]
 
         [0160]    Then, the discontinuity detector  129  detects a discontinuity, for example, by the below equation (step S 26  in  FIG. 15 ). 
         [0000]        E   0   −E   −1 &gt;Thres  [Mathematical Equation 66]
 
         [0161]    When the gain change does not exceed the threshold (NO in step S 27  of  FIG. 15 ), the discontinuity detector  129  does not detect an occurrence of a discontinuity, and the ISP-LPC converter  122 C outputs the LP coefficients and ends the processing. On the other hand, when the gain change exceeds the threshold (YES in step S 27  of  FIG. 15 ), the discontinuity detector  129  detects an occurrence of a discontinuity and sends a control signal indicative of a result of the detection for an occurrence of a discontinuity to the stability processor  121 . When receiving the control signal, the stability processor  121  corrects the ISP parameters in the same manner as performed in the first embodiment (step S 28  in  FIG. 15 ). The subsequent operations of the ISF-ISP converter  122 A, ISP interpolator  122 B, and ISP-LPC converter  122 C (steps S 29 , S 2 A, and S 2 B in  FIG. 15 ) are the same as above. 
         [0162]    As discussed in the above second embodiment, a discontinuity of decoded audio can be determined by the power of the excitation signal, and the discontinuous audio is reduced to improve the subjective quality in the same manner as performed in the first embodiment. 
       Third Embodiment 
       [0163]    Upon a detection of discontinuity, the ISF parameters may be corrected by another method. The third embodiment differs from the first embodiment only in the stability processor  121 , and thus only the operation of the stability processor  121  will be described. 
         [0164]    When the discontinuity detector  129  detects a discontinuity, the stability processor  121  performs the following process to correct the ISF parameters. 
         [0165]    With respect to the ISF parameters stored in the internal state buffer  14 , 
         [0000]      {dot over (ω)} i   −1   [Mathematical Equation 67]
 
         [0000]    the stability processor  121  replaces the ISF parameters up to a low-order P′ dimension (0&lt;P′≦P) in accordance with the below equation. Here, the following definition is adopted. 
         [0000]      δ −1 ={dot over (ω)} P′−1   −1   /P′   [Mathematical Equation 68]
 
         [0000]      {dot over (ω)} i   −1 ={dot over (ω)} i−1   −1 +δ −1  
 
         [0000]      {dot over (ω)} 0   −1 =δ −1 (0 ≦i&lt;P′ )  [Mathematical Equation 69]
 
         [0166]    The stability processor  121  may overwrite the ISF parameters of the low-order P′ dimensions with P′-dimension vectors obtained in advance by learning as follows. 
         [0000]      {dot over (ω)} i   −1 =ω i   0 (0 ≦i&lt;P′ )  [Mathematical Equation 70]
 
         [0167]    Next, as to the ISF parameters of the current frame, the stability processor  121  may, as performed in the first embodiment, perform the process of expanding the distance between elements to become M 0  times wider than the ordinary distance or may determine them in accordance with the below equation. Here, the following definition is adopted. 
         [0000]      δ 0 ={dot over (ω)} P′−1   0   /P′   [Mathematical Equation 71]
 
         [0000]      {dot over (ω)} i   0 ={dot over (ω)} i−1   0 +δ 0  
 
         [0000]      {dot over (ω)} 0   0 =δ 0   [Mathematical Equation 72]
 
         [0168]    The stability processor  121  may overwrite them with P′-dimensional vectors learned in advance. 
         [0000]      {dot over (ω)} i   0 =ω i   0 (0 ≦i&lt;P′ )  [Mathematical Equation 73]
 
         [0169]    Furthermore, the foregoing P′-dimensional vectors may be learned in the decoding process or may be defined, for example, as follows. 
         [0000]      ω i   0 =(1−λ)ω i   −1 +λ{dot over (ω)} i   −1   [Mathematical Equation 74]
 
         [0000]    In a frame at the start of decoding, however, ω −1  may be defined as predetermined P′-dimensional vector ω i   init . 
         [0170]    The internal state buffer  14  updates the ISF parameters stored in the past with the new ISF parameters. 
         [0171]    As discussed in the above third embodiment, the distance obtained by equally dividing the ISF/LSF parameters into those of a predetermined dimension can be used as the distance between elements of the ISF/LSF parameters given for ensuring the stability of the synthesis filter, whereby the discontinuous audio is reduced to improve the subjective quality as performed in the first and second embodiments. 
       Fourth Embodiment 
       [0172]    A fourth embodiment will be described in which the encoding side detects an occurrence of a discontinuity and transmits a discontinuity determination code (indicative of a detection result) as included in audio codes to the decoding side and also in which the decoding side determines the operation of the stability process, based on the discontinuity determination code included in the audio codes. 
         [0173]    (Regarding Encoding Side) 
         [0174]      FIG. 16  shows an exemplary functional configuration of the encoder  2 , and  FIG. 17  is a flowchart showing the processes performed in the encoder  2 . As shown in  FIG. 16 , the encoder  2  has an LP analyzer/encoder  21 , a residual encoder  22 , and a code multiplexer  23 . 
         [0175]    An exemplary functional configuration of the LP analyzer/encoder  21  among them is shown in  FIG. 18 , and a flowchart showing the processes performed in the LP analyzer/encoder  21  is shown in  FIG. 19 . As shown in  FIG. 18 , the LP analyzer/encoder  21  has an LP analyzer  210 , an LP-ISF converter  211 , an ISF encoder  212 , a discontinuity determiner  213 , an ISF concealer  214 , an ISF-LP converter  215 , and an ISF buffer  216 . 
         [0176]    In the LP analyzer/encoder  21 , the LP analyzer  210  performs a linear prediction analysis on an input signal to obtain linear prediction coefficients (step T 41  in  FIG. 17  and step U 41  in  FIG. 18 ). For the calculation of linear prediction coefficients, an autocorrelation function is first calculated from the audio signal, and then the Levinson-Durbin algorithm or the like can be applied. 
         [0177]    The LP-ISF converter  211  converts the calculated linear prediction coefficients into the ISP parameters in the same manner as performed in the first embodiment (steps T 42 , U 42 ). The conversion from linear prediction coefficients into ISF parameters may be implemented by use of the method described in the Non Patent Literature. 
         [0178]    The ISF encoder  212  encodes the ISF parameters using a predetermined method to calculate ISF codes (steps T 43 , U 43 ) and outputs quantized ISF parameters obtained in the process of encoding to the discontinuity determiner  213 , the ISF concealer  214 , and the ISF-LP converter  215  (step U 47 ). Here, the quantized ISF parameters are equal to the ISF parameters obtained by an inverse quantization of the ISF codes. A method of encoding may be vector-encoding, or encoding by a vector quantization or the like of error vectors from ISFs of the immediately preceding frame and mean vectors determined in advance by learning. 
         [0179]    The discontinuity determiner  213  encodes a discontinuity determination flag stored in an internal buffer (not shown) built in the discontinuity determiner  213  and outputs a resultant discontinuity determination code (step U 47 ). In addition, the discontinuity determiner  213  uses concealment ISF parameters 
         [0000]      {tilde over (ω)} i   [Mathematical Equation 75]
 
         [0000]    read from the ISF buffer  216  and the quantized ISF parameters 
         [0000]      {dot over (ω)} i   [Mathematical Equation 76]
 
         [0000]    to make a determination on a discontinuity in accordance with the below equation (steps T 44 , U 46 ). Here, Thres ω  represents a threshold determined in advance, and P′ an integer satisfying the following equation (0&lt;P′≦P). 
         [0000]      Σ i=0   P′−1 ({dot over (ω)} i −{tilde over (ω)} i ) 2 &gt;Thres ω   [Mathematical Equation 77]
 
         [0180]    The example is described above in which the discontinuity determination is made using the Euclidean distances between the ISF parameters. However, the discontinuity determination may be made by other methods. 
         [0181]    The ISF concealer  214  calculates the concealment ISF parameters from the quantized ISF parameters by the same process as performed by the decoder-side ISF concealer and outputs the resultant concealment  1 SF parameters to the ISF buffer  216  (steps U 44 , U 45 ). The operation of the ISF concealment process may be performed by any method as long as it is the same process as that of the decoder-side packet loss concealer. 
         [0182]    The ISF-LP converter  215  calculates quantized linear prediction coefficients by converting the foregoing quantized ISF parameters and outputs a resultant quantized linear prediction coefficients to the residual encoder  22  (step T 45 ). A method used for converting the ISF parameters into the quantized linear prediction coefficients may be the method described in the Non Patent Literature. 
         [0183]    The residual encoder  22  filters the audio signal by use of the quantized liner prediction coefficients to calculate residual signals (step T 46 ). 
         [0184]    Next, the residual encoder  22  encodes the residual signals by encoding means using CELP or TCX (Transform Coded Excitation) or by encoding means switchably using CELP and TCX and outputs resultant residual codes (step T 47 ). Since the operation of the residual encoder  22  is less relevant to the present invention, description thereof is omitted herein. 
         [0185]    The code multiplexer  23  assembles the ISF codes, the discontinuity determination code and the residual codes in a predetermined order and outputs resultant audio codes (step T 48 ). 
         [0186]    (Regarding Decoding Side) 
         [0187]    An audio signal processing device according to the fourth embodiment has the same configuration as that of the aforementioned audio decoder  1  in  FIG. 1  and has a novel feature in the audio code decoder, and thus the audio code decoder will be described below.  FIG. 20  shows an exemplary functional configuration of an audio code decoder  12 D, and  FIG. 21  is a flowchart showing the process of calculating the LP coefficients. The audio code decoder  12 D shown in  FIG. 20  is configured by adding the discontinuity detector  129  to the aforementioned configuration shown in  FIG. 3 . 
         [0188]    The ISF decoder  120  decodes the ISF codes and outputs resultant codes to the stability processor  121  and the internal state buffer  14  (step S 41  in  FIG. 21 ). 
         [0189]    The discontinuity detector  129  decodes the discontinuity determination code and outputs a resultant discontinuity detection result to the stability processor  121  (step S 42  in  FIG. 21 ). 
         [0190]    The stability processor  121  performs the stability process according to the discontinuity detection result (step S 43  in  FIG. 21 ). The processing procedure of the stability processor to be used can be the same method as executed in the first embodiment and the third embodiment. 
         [0191]    The stability processor  121  may perform the stability process as described below, on the basis of other parameters included in the audio codes, in addition to the discontinuity detection result acquired from the discontinuity determination code. For example, the stability processor  121  may be configured to perform the stability process in such a manner that an ISF stability stab is calculated in accordance with the below equation and that when the ISF stability exceeds a threshold, even if the discontinuity determination code shows a detection of a discontinuity, the process is performed as if no discontinuity is detected. Here, C is a constant determined in advance. 
         [0000]      stab=1.25−Σ i=0   P′−1 ({dot over (ω)} i   0 −{dot over (ω)} i   −1 ) 2   /C   [Mathematical Equation 78]
 
         [0192]    The ISF-ISP converter  122 A in the LP coefficient calculator  122  converts the ISF parameters into the ISP parameters by the same processing procedure as performed in the first embodiment (step S 44  in  FIG. 21 ). 
         [0193]    The ISP interpolator  122 B calculates the ISP parameters for the respective subframes by the same processing procedure as performed in the first embodiment (step S 45  in  FIG. 21 ). 
         [0194]    The ISP-LPC converter  122 C converts the ISP parameters calculated for the respective subframes into the LPC parameters by the same processing procedure as performed in the first embodiment (step S 46  in  FIG. 21 ). 
         [0195]    In the fourth embodiment as described above, the encoding side performs the discontinuity determination (the discontinuity determination using the Euclidian distances between concealment ISF parameters and quantized ISF parameters, as an example) encodes auxiliary information about a result of the determination and outputs encoded information to the decoding side, and the decoding side determine a discontinuity using the auxiliary information obtained by decoding. In this manner, the appropriate processing can be executed according to the discontinuity determination result made by the encoding side while the encoding side and the decoding side work in concert with each other. 
       Fifth Embodiment 
       [0196]    (Regarding Encoding Side) 
         [0197]    The functional configuration of the encoder is the same as that of the fourth embodiment shown in  FIG. 16 , and the processing flow of the encoder is the same as the processing flow of the fourth embodiment shown in  FIG. 17 . The below will describe the LP analyzer/encoder according to the fifth embodiment which is different from that in the fourth embodiment. 
         [0198]      FIG. 22  shows an exemplary functional configuration of the LP analyzer/encoder, and  FIG. 23  shows a flow of the processes performed by the LP analyzer/encoder. As shown in  FIG. 22 , the LP analyzer/encoder  21 S has the LP analyzer  210 , the LP-ISF converter  211 , the ISF encoder  212 , the discontinuity determiner  213 , the ISF concealer  214 , the ISF-LP converter  215 , and the ISF buffer  216 . 
         [0199]    In this LP analyzer/encoder  21 S, the LP analyzer  210  performs the linear prediction analysis on the input signal by the same process as performed in the fourth embodiment to obtain the linear prediction coefficients (step U 51  in  FIG. 23 ). 
         [0200]    The LP-ISF converter  211  converts the calculated linear prediction coefficients into the ISF parameters by the same process as performed in the fourth embodiment (step U 52  in  FIG. 23 ). The method described in the Non Patent Literature may be used for the conversion from the linear prediction coefficients into the ISF parameters. 
         [0201]    The ISF encoder  212  reads the discontinuity determination flag stored in the internal buffer (not shown) of the discontinuity determiner  213  (step U 53  in  FIG. 23 ). 
         [0202]    &lt;Case Where Discontinuity Determination Flag Indicates Detection of Discontinuity&gt; 
         [0203]    The ISF encoder  212  calculates the ISF codes by vector-quantization of ISF residual parameters r i  calculated by the below equation (step U 54  in  FIG. 23 ). Here, the ISF parameters calculated by the LP-ISF converter are denoted by ω i  and mean vectors, which are mean i , obtained in advance by learning. 
         [0000]        r   i =ω i −mean i   [Mathematical Equation 79]
 
         [0204]    Next, the ISF encoder  212  uses the quantized ISF residual parameters 
         [0000]      {circumflex over (r)} i   [Mathematical Equation 80]
 
         [0000]    obtained by quantization of the ISF residual parameters r i  to update the ISF residual parameter buffer in accordance with the following equation (step U 55  in  FIG. 23 ). 
         [0000]        {dot over (r)}   −1   ={circumflex over (r)}   i   [Mathematical Equation 81]
 
         [0205]    &lt;Case Where Discontinuity Determination Flag does not Indicate Detection of Discontinuity&gt; 
         [0206]    The ISF encoder  212  calculates the ISF codes by vector-quantization of the ISF residual parameters r, calculated by the below equation (step U 54  in  FIG. 23 ). Here, the ISF residual parameters obtained by decoding in the immediately preceding frame are denoted as follows. 
         [0000]      {dot over (r)} i   −1   [Mathematical Equation 82]
 
         [0000]        r   i =ω i −mean i −⅓ {dot over (r)}   i   −1   [Mathematical Equation 83]
 
         [0207]    Next, the ISF encoder  212  uses the quantized ISF residual parameters 
         [0000]      {circumflex over (r)} i   [Mathematical Equation 84]
 
         [0000]    obtained by quantization of the ISF residual parameters r i  to update the ISF residual parameter buffer in accordance with the following equation (step U 55  in  FIG. 23 ). 
         [0000]        {dot over (r)}   i   −1   ={circumflex over (r)}   i   [Mathematical Equation 85]
 
         [0208]    By the above procedure, the ISF encoder  212  calculates the ISF codes and outputs quantized ISF parameters obtained in the process of encoding to the discontinuity determiner  213 , the ISF concealer  214 , and the ISF-LP converter  215 . 
         [0209]    The ISF concealer  214  calculates the concealment ISF parameters from the quantized ISF parameters by the same process as performed by the decoder-side ISF concealer in the same manner as executed in the fourth embodiment and outputs them to the  1 SF buffer  216  (steps U 56 , U 58  in  FIG. 23 ). The operation of the ISF concealment process may be performed by any method as long as it is the same process as that of the decoder-side packet loss concealer. 
         [0210]    The discontinuity determiner  213  performs a determination of a discontinuity by the same process as performed in the fourth embodiment and stores a determination result in the internal buffer (not shown) of the discontinuity determiner  213  (step U 57  in  FIG. 23 ). 
         [0211]    The ISF-LP converter  215  converts the quantized ISF parameters, in the same manner as performed in the fourth embodiment, to calculate the quantized linear prediction coefficients and outputs them to the residual encoder  22  ( FIG. 16 ) (step U 58  in  FIG. 23 ). 
         [0212]    (Regarding Decoding Side) 
         [0213]    An audio signal processing device according to the fifth embodiment has the same configuration as that of the aforementioned audio decoder  1  in  FIG. 1  and has a novel feature in the audio code decoder, and thus the audio code decoder will be described below.  FIG. 24  shows an exemplary functional configuration of the audio code decoder  12 E, and  FIG. 25  shows a flow of the calculation process performed by the LP coefficients. The audio code decoder  12 E shown in  FIG. 24  is configured by adding the discontinuity detector  129  to the aforementioned configuration shown in  FIG. 3 . 
         [0214]    The discontinuity detector  129  decodes the discontinuity determination code and outputs the resultant discontinuity determination flag to the ISF decoder  120  (step S 51  in  FIG. 25 ). 
         [0215]    The ISF decoder  120  calculates the ISF parameters as follows, depending upon the value of the discontinuity determination flag, and outputs the ISF parameters to the stability processor  121  and the internal state buffer  14  (step S 52  in  FIG. 25 ). 
         [0216]    &lt;Case Where Discontinuity Determination Flag Indicates Detection of Discontinuity&gt; 
         [0217]    The ISF decoder  120  uses the quantized ISF residual parameters 
         [0000]      {dot over (r)} i   [Mathematical Equation 86]
 
         [0000]    obtained by decoding of the ISF codes, and the mean vectors mean i  obtained in advance by learning to obtain the quantized ISF parameters 
         [0000]      {dot over (ω)} i   [Mathematical Equation 87]
 
         [0000]    in accordance with the following equation. 
         [0000]      {dot over (ω)} i mean i   +{dot over (r)}   i   [Mathematical Equation 88]
 
         [0218]    Next, the ISF decoder  120  updates the ISF residual parameters stored in the internal state buffer  14  in accordance with the following equation. 
         [0000]      {dot over (r)} i   −1   ={dot over (r)}   i   [Mathematical Equation 89]
 
         [0219]    &lt;Case Where Discontinuity Determination Flag Does Not Indicate Detection of Discontinuity&gt; 
         [0220]    The ISF decoder  120  reads, from the internal state buffer  14 , the ISF residual parameters 
         [0000]      {dot over (r)} i   −1   [Mathematical Equation 90]
 
         [0000]    obtained by decoding of the immediately preceding frame and uses the resultant ISF residual parameters 
         [0000]      {dot over (r)} i   −1 ,  [Mathematical Equation 91]
 
         [0000]    the mean vectors mean i  obtained in advance by learning and the quantized ISF residual parameters 
         [0000]      {dot over (r)} i   [Mathematical Equation 92]
 
         [0000]    obtained by decoding of the ISF codes to calculate the quantized ISF parameters 
         [0000]      {dot over (ω)} i   [Mathematical Equation 93]
 
         [0000]    in accordance with the following equation. 
         [0000]      {dot over (ω)} i =mean i   +{dot over (r)}   i +⅓ {dot over (r)}   i   −1   [Mathematical Equation 94]
 
         [0221]    Next, the ISF decoder  120  updates the ISF residual parameters stored in the internal state buffer  14  in accordance with the following equation. 
         [0000]        {dot over (r)}   i   −1   ={dot over (r)}   i   [Mathematical Equation 95]
 
         [0222]    The stability processor  121  performs the same process as performed in the first embodiment (step S 53  in  FIG. 25 ) when a discontinuity is not detected. 
         [0223]    The ISF-ISP converter  122 A in the LP coefficient calculator  122  converts the ISF parameters into the ISP parameters by the same processing procedure as described in the first embodiment (step S 54  in  FIG. 25 ). 
         [0224]    The ISP interpolator  122 B calculates the ISP parameters for the respective subframes by the same processing procedure as performed in the first embodiment (step S 55  in  FIG. 25 ). 
         [0225]    The ISP-LPC converter  122 C, by the same processing procedure as performed in the first embodiment (step S 56  in  FIG. 25 ), converts the ISP parameters calculated for the respective subframes into the LPC parameters. 
         [0226]    In the fifth embodiment as described above, the encoding side is configured as follows: When the discontinuity determination flag does not indicate a detection of a discontinuity, the vector quantization of the ISF residual parameters is carried out using the ISF residual parameters obtained by decoding of the immediately preceding frame. On the other hand, when the discontinuity determination flag indicates a detection of a discontinuity, the encoder avoids using the ISF residual parameters obtained by decoding of the immediately preceding frame. Similarly, the decoding side is configured as follows: When the discontinuity determination flag does not indicate a detection of a discontinuity, the quantized ISF parameters are calculated using the ISF residual parameters obtained by decoding of the immediately preceding frame. On the other hand, when the discontinuity determination flag indicates a detection of discontinuity, the decoder avoids using the ISF residual parameters obtained by decoding of the immediately preceding frame. In this manner, the appropriate processing according to a discontinuity determination result can be executed while the encoding side and the decoding side work in concert with each other. 
       Sixth Embodiment 
       [0227]    The above first to fifth embodiments may be applied in combination. For example, as described in the fourth embodiment, the decoding side decodes the discontinuity determination code included in the audio codes from the encoding side to detect a discontinuity. When a discontinuity is detected, it may carry out the subsequent operation as follows. 
         [0228]    For the ISF parameters 
         [0000]      {dot over (ω)} i   −1   [Mathematical Equation 96]
 
         [0000]    stored in the internal state buffer, the ISF parameters up to the low-degree P′ dimension (0&lt;P′≦P) are replaces in accordance with the following equation as described in the third embodiment. 
         [0000]      {dot over (ω)} i   −1 =ω i   0 (0 ≦i&lt;P′ )  [Mathematical Equation 97]
 
         [0229]    On the other hand, the ISF parameters of the current frame are calculated in accordance with the following equation as described in the fifth embodiment. 
         [0000]      {dot over (ω)} i =mean i   +{dot over (r)}   i   [Mathematical Equation 98]
 
         [0230]    Thereafter, using the ISF parameters obtained as described above, the LP coefficients are obtained by the processes of the ISF-ISP converter  122 A, the ISP interpolator  122 B, and the ISP-LPC converter  122 C as performed in the first embodiment. 
         [0231]    It is also effective to adopt optional combinations of the first to fifth embodiments as described above. 
       Seventh Embodiment 
       [0232]    It may be considered in the decoding operation according to the above first to sixth embodiments and their modifications, how the frame is lost (e.g., whether a single frame is lost or consecutive frames are lost). In the seventh embodiment, it suffices that a discontinuity detection is made using, for example, the result of decoding of the discontinuity determination code included in the audio codes, and the method of how it should be performed is not limited to the above. 
         [0233]    An audio signal processing device according to the seventh embodiment has the same configuration as that of the aforementioned audio decoder  1  in  FIG. 1  and has a novel feature in the audio code decoder, and thus the audio code decoder will be described below. 
         [0234]      FIG. 26  shows an exemplary configuration of the audio decoder  1 S according to the seventh embodiment, and  FIG. 27  shows a flowchart of the processes performed in the audio decoder. As shown in  FIG. 26 , in addition to the aforementioned audio code decoder  12 G, the concealment signal generator  13  and the internal state buffer  14 , the audio decoder  1 S has a reception state determiner  16  that determines packet reception states in some past frames and stores a packet loss history. 
         [0235]    The reception state determiner  16  determines a packet reception state and updates the packet loss history information, based on a determination result (step S 50  in  FIG. 27 ). 
         [0236]    When a packet loss is detected (NO in step S 100 ), the reception state determiner  16  outputs a packet loss detection result of the pertinent frame to the concealment signal generator  13 , and the concealment signal generator  13  generates the concealment signal as described above and updates the internal states (steps S 300 , S 400 ). The concealment signal generator  13  may also utilize the packet loss history information for interpolation of parameters or the like. 
         [0237]    On the other hand, when no packet loss is detected (YES in step S 100 ), the reception state determiner  16  outputs the packet loss history information including a packet loss detection result of the pertinent frame and the audio codes included in the received packet to the audio code decoder  12 , and the audio code decoder  12  decodes the audio codes as described before and updates the internal states (steps S 200 , S 400 ). 
         [0238]    Thereafter, the processes of steps S 50  to S 400  are repeated until the communication ends (or until step S 500  results in a determination of YES). 
         [0239]      FIG. 28  shows an exemplary functional configuration of the audio code decoder  12 G, and  FIG. 29  shows a flowchart of the calculation processes performed by the LP coefficients. An example will be described below using the packet loss history information only for the LP coefficient calculator  122 , but the audio code decoder may be configured to use the packet loss history information for other constitutive elements. 
         [0240]    Since the audio code decoder  12 G has the same configuration as described in the first embodiment, except for the configuration associated with the calculation process of LP coefficients, the below will describe the configuration and its operation associated with the calculation process of LP coefficients. 
         [0241]    The ISF decoder  120  decodes the ISF codes in the same manner as performed in the first embodiment and outputs the ISF parameters to the stability processor  121  (step S 71  in  FIG. 29 ). 
         [0242]    The discontinuity detector  129  refers to the packet loss history information to determine the reception state (step S 72 ). The discontinuity detector  129  may be designed, for example, as follows: It stores a specific reception pattern which indicates, for example, a packet loss occurred three frames prior, a normal reception occurred two frames prior, and a packet loss occurred one frame prior. When the reception pattern is recognized which has been looked for, it sets a reception state flag to off and, otherwise, it sets the reception state flag to on. 
         [0243]    Furthermore, the discontinuity detector  129  detects a discontinuity in the same manner as described in one of the first to sixth embodiments. 
         [0244]    Then, the stability processor  121  performs the stability process according to the reception state flag and a result of the discontinuity detection, for example, as described below (step S 73 ). 
         [0245]    When the reception state flag is off, the stability processor  121  performs the same process as performed when a discontinuity is not detected, regardless of a result of the discontinuity detection. 
         [0246]    On the other hand, when the reception flag is on and when the result of the discontinuity detection indicates that a discontinuity is not detected, the stability processor  121  performs the same process as performed when a discontinuity is not detected. 
         [0247]    Furthermore, when the reception flag is on and when the result of the discontinuity detection is detection of discontinuity, the stability processor  121  performs the same process as performed when a discontinuity is detected. 
         [0248]    Thereafter, the operations (steps S 74  to S 76 ) of the ISF-ISP converter  122 A, the ISP interpolator  122 B, and the ISP-LPC converter  122 C in the LP coefficient calculator  122  are performed in the same manners as performed in the first embodiment. 
         [0249]    In the seventh embodiment as described above, the stability process is carried out depending upon a result of the discontinuity detection and the state of the reception state flag, whereby more accurate processing can be executed while it is considered how the frame is lost (e.g., whether a single frame is lost or consecutive frames are lost). 
         [0250]    [Regarding Audio Signal Processing Programs] 
         [0251]    The below will describe audio signal processing programs that program a computer to operate as an audio signal processing device according to the present invention. 
         [0252]      FIG. 32  is a drawing showing various exemplary configurations of the audio signal processing programs.  FIG. 30  is an exemplary hardware configuration of the computer, and  FIG. 31  shows a schematic view of a computer. Audio signal processing programs P 1 -P 4  (which will be referred to hereinafter generally as “audio signal processing program P”) shown in  FIG. 32 ( a ) to ( d ) , respectively, can program the computer C 10  shown in  FIGS. 31 and 32  to operate as an audio signal processing device. It should be noted that the audio signal processing program P described in the present specification can be implemented not only on the computer as shown in  FIGS. 31 and 32  but also on any information processing device such as a cell phone, a personal digital assistance, or a portable personal computer. 
         [0253]    The audio signal processing program P can be provided in a form stored in a recording medium M. Examples of the recording medium M include recording media such as flexible disc, CD-ROM, DVD, or ROM, semiconductor memories, and so on. 
         [0254]    As shown in  FIG. 30 , the computer C 10  has a reading device C 12  such as a flexible disc drive unit, a CD-ROM drive unit, or a DVD drive unit, a working memory (RAM) C 14 , a memory C 16  for storing a program stored in the recording medium M, a display C 16 , a mouse C 20  and a keyboard C 22  as input devices, a communication device C 24  for executing transmission/reception of data or the like, and a central processing unit (CPU) C 26  for controlling execution of the program. 
         [0255]    When the recording medium M is put into the reading device C 12 , the computer C 10  becomes accessible to the audio signal processing program P stored in the recording medium M through the reading device C 12  and becomes able to operate as an audio signal processing device programmed by the audio signal processing program P. 
         [0256]    The audio signal processing program P may be one provided as computer data signal W superimposed on a carrier wave, as shown in  FIG. 31 , transmitted through a network. In this case, the computer C 10  stores the audio signal processing program P received by the communication device C 24  into the memory C 16  and then can execute the audio signal processing program P. 
         [0257]    The audio signal processing program P can be configured by adopting the various configurations shown in  FIG. 32 ( a ) to ( d ) . These correspond to the configurations recited in claims  18  to  21  associated with the audio signal processing programs as set forth in the scope of claims. For example, the audio signal processing program P 1  shown in  FIG. 32 ( a )  has a discontinuity detection module P 11  and a discontinuity correction module P 12 . The audio signal processing program P 2  shown in  FIG. 32 ( b )  has an ISF/LSF quantization module P 21 , an ISF/LSF concealment module P 22 , a discontinuity detection module P 23 , and an auxiliary information encoding module P 24 . The audio signal processing program P 3  shown in  FIG. 32 ( c )  has a discontinuity detection module P 31 , an auxiliary information encoding module P 32 , and an ISF/LSF quantization module P 33 . The audio signal processing program P 4  shown in  FIG. 32 ( d )  has an auxiliary information decoding module P 41 , a discontinuity correction module P 42 , and an ISF/LSF decoding module P 43 . 
         [0258]    By implementing the various embodiments described above, the subjective quality can be improved while reducing a discontinuous audio which can occur in the recovery from a packet loss at the audio start point. 
         [0259]    The stability processor, which is the first feature of the invention, is configured so that when a discontinuity is detected in the first packet which is received correctly after a packet loss occurs, for example, a distance between elements of the  1 SF parameters is set wider than normal, whereby it can prevent the gain of the LP coefficients from becoming too large. Since it can prevent both the gain of the LP coefficient and the power of the excitation signal from increasing, a discontinuity of the synthesized signal is reduced, whereby a degradation of the subjective quality can be suppressed. Furthermore, the stability processor may reduce a discontinuity of the synthesized signal by multiplying the synthesized signal by the gain calculated by using the LP coefficients or the like. 
         [0260]    The discontinuity detector, which is the second feature of the invention, monitors the gain of the excitation signal included in the first packet which is received correctly after a packet loss occurs, and determines a discontinuity for a packet whose gain of the excitation signal increased more than a certain level.