Abstract:
A speech decoding device is provided which is capable of reducing degradation of speech quality caused by concealment processing to be performed when a loss of a packet has occurred, in speech packet communications using a VoIP (Voice over Internet Protocol) or a like. A decoding circuit decodes speech from a packet received through an input terminal and stores an internal signal in an updating buffer circuit, the internal signal produced in the decoding process and to be used in a decoding process for a subsequent packet to be subsequently received. The decoding circuit produces, based on the internal signal stored in the updating buffer circuit, concealed speech corresponding to a packet having not been received, and outputs the produced concealed speech. At this point, the updating circuit, by regarding the concealed speech produced by the decoding circuit as being not differing greatly from an original speech and by updating the internal signal stored in the updating buffer circuit using the concealed speech, reduces mismatching of internal signals occurring due to the concealment processing between in an encoding device and in a decoding device and reduces degradation of speech quality in decoding a packet following the concealment processing.

Description:
BACKGROUND OF THE INVENTION  
         [0001]    1. Field of the Invention  
           [0002]    The present invention relates to a speech decoding device and a speech decoding method, and more particularly to the speech decoding device and the method for decoding speech being capable of reducing degradation of speech quality caused by concealment processing to be performed when a loss of a packet has occurred, in speech packet communications using a VoIP (Voice over Internet Protocol) or a like.  
           [0003]    The present application claims priority of Japanese Patent Application No. 2002-117187 filed on Apr. 19, 2002, which is hereby incorporated by reference.  
           [0004]    2. Description of the Related Art  
           [0005]    In packet-type speech communications such as a VoIP (Voice over Internet Protocol) system or a like, a transmitter combines one piece of speech frame data or a plurality of pieces of speech frame data obtained by encoding speech in a block unit of 10 msec or a like into one packet and, after having added information such as a produced time or a like to the packet, transmits it through a transmission path including the Internet or a like.  
           [0006]    In the transmission path, a transmitted packet reaches a receiver through a plurality of repeaters such as a router, gateway, or a like. Since a packet is stored in a queue while passing through the repeater, there are some cases in which, if the repeater is put in a busy state, the packet is re-transmitted after much time has elapsed since its receipt or the packet is discarded due to no processing by the repeater in time. The receiver judges whether or not an order or a time given to a time stamp added to received packets is in compliance with predetermined rules. If it is not in compliance with the predetermined rules, the packet is regarded as lost. By using a concealment process to be performed on a portion corresponding to a lost packet, speech corresponding to the lost packet is decoded.  
           [0007]    In the above concealment process, though its process varies depending on a method of encoding speech to be applied, based on information contained in packets having received before or after the lost packet, speech corresponding to the lost packet is produced. When a packet having been transmitted after the lost packet is used for the concealment process, a delay in decoding occurs because of receiving process of the packet.  
           [0008]    A concealment process according to a CELP (Code Excited Linear Prediction) method being employed in various types of portable cellular phones is described, for example, in “Performance of the Proposed ITU-T 8 kb/s Speech Coding Standard for a Rayleigh Fading Channel” (IEEE Proc. Speech Coding Workshop, pp. 11-12, 1995) (Reference No. 1). A concealment process according to an ADPCM (Adaptive Differential Pulse Code Modulation) method being employed in a PHS (Personal Handy-Phone System) is described, for example, in “Improved ADPCM Voice Signal Transmission Employing Click-Noise Detection Scheme for TDMA-TDD Personal Communication System” (IEEE Trans. On Vehicular Technology, Vol. 46, No. 1, 1997) (Reference No. 2). Moreover, a same concealment process as used in the above ADPCM method can be applied to a band-splitting-type ADPCM method in which speech in a wide band of up to 7 kHz is encoded.  
           [0009]    Examples of configurations of a conventional speech decoding device in which a packet loss concealment process is performed are explained by referring to FIGS. 9, 10,  11 , and  12 . FIG. 9 is a schematic block diagram showing an entire configuration of the conventional speech decoding device. FIGS. 10, 11, and  12  are schematic block diagrams illustrating speech decoding circuits employed in the conventional speech decoding device. That is, FIG. 10 is a block diagram showing an all-band-type decoding circuit to decode speech in all bands by using the CELP method and FIG. 11 is a block diagram showing an all-band-type decoding circuit to decode speech in all bands by using the ADPCM method. FIG. 12 is a block diagram showing a band-splitting-type decoding circuit to produce all band signals by performing an addition on signals obtained by splitting a band to decode speech.  
           [0010]    Operations of the conventional speech decoding device are described by referring to FIG. 9. An input terminal  15  receives a packet and passes it to a decoding circuit  30 . The input terminal  15  receives loss information indicating whether or not there is a loss of a packet and passes the information to the decoding circuit  30 . The decoding circuit  30  decodes speech from packets fed from the input terminal  15  according to the loss information fed from an input terminal  10 . Moreover, when speech is decoded from each of packets, an internal signal contained in a previous packet fed from a buffer circuit  35  is used. Then, after the decoding, the internal signal contained in the previous packet to be used in decoding a subsequent packet is passed to the buffer circuit  35 . The internal signal to be used varies depending on a speech encoding method. Concrete examples of the decoding circuit  30  will be explained later by referring to FIGS. 10 and 11. Finally, decoded speech is passed to an output terminal  45 . The buffer circuit  35  stores the internal signal fed from the decoding circuit  30  and passes the internal signals that had been stored at a time of speech decoding from a subsequent packet to the decoding circuit  30 . The output terminal  45  outputs the decoded speech fed from the decoding circuit  30 .  
           [0011]    [0011]FIG. 10 is a block diagram showing an example of a conventional decoding circuit employed in a decoding device using the CELP method, in which the decoding circuit  30  shown in FIG. 9 is provided as a decoding circuit  203  in FIG. 10. The CELP method is described in “Code—Excited Linear Prediction: High Quality Speech at Very Low Bit Rates (IEEE Proc. ICASSP-85, pp. 937-940, 1985) (Reference No. 3). In the encoding device operated according to the CELP method, input speech is split into a linear prediction (LP) coefficient portion showing a spectrum enveloping characteristic obtained by a linear prediction (LP) analysis and an exciting signal used to drive an LP synthetic filter made up of the above LP coefficient portion to perform encoding. The LP analysis and encoding of the LP coefficient portion are performed for every frame having a predetermined length. Encoding of the exciting signal is performed for every sub-frame having a predetermined length obtained by further dividing the frame. Here, the exciting signal is made up of a pitch component representing a pitch period, a residual component other than the pitch component and a gain of each of the these components. The pitch component representing a pitch period of an input signal is expressed by an adaptive code vector stored in a code book called an “adaptive code book” holding exciting signals received in the past. The above residual component is expressed by a signal designed in advance called a “speech source code vector”. As this signal, a multi-pulse signal made up of a plurality of pulses, a random number signal, or a like are used. Information about a speech source code vector is stored in a speech source code book. In the CELP-type decoding device, by inputting an exciting signal calculated from the decoded pitch period component and the residual signal into a synthetic filter made up of the decoded LP coefficient portion to calculate decoded speech.  
           [0012]    Next, operations of the decoding circuit 203 (CELP-type) are described by referring to FIG. 10. In this specification, to simplify descriptions, a case where one frame is contained in one packet is described, however, even if a plurality of frames is contained in one packet, decoding is made possible by repeating operations in a same manner as described below. An input terminal  50  receives a packet and passes it to a speech source analyzing circuit  65 , a pitch predicting circuit  68 , and a synthetic filter circuit  88 . An input terminal  55  receives loss information and passes it to the synthetic filter circuit  88 , the speech source analyzing circuit  65 , and the pitch predicting circuit  68 . The speech source analyzing circuit  65  decodes a speech source code vector and its gain by using information indicated by a packet fed from the input terminal  50  and passes a speech source signal obtained by adding up the speech source code vector and its gain to an adder  75 . However, if the loss information fed from the input terminal  55  indicates occurrence of loss of a packet, the speech source analyzing circuit  65  produces a pseudo speech source signal such as a random number or a like and passes it to the adder  75 . The pitch predicting circuit  68  decodes an adaptive code vector and its gain by using information indicated by the packet fed from the input terminal  50  and passes a pitch period signal obtained by adding up the adaptive code vector and its gain to the adder  75 . The adaptive code vector is obtained by allocating the adaptive code vector being stored as an internal signal from the buffer circuit  35  being placed outside and being connected through an input/output terminal  80 . If the loss information fed from the input terminal  55  indicates occurrence of loss of a packet, a signal made up of, for example, “zero” is passed to the adder  75  as a pitch period signal. The adder  75  feeds an exciting signal obtained by adding up a speech source signal fed from the speech source analyzing circuit  65  and a pitch period signal fed from the pitch predicting circuit  68  to the synthetic filter circuit  88  and, at a same time, passes it as an internal signal through the input/output terminal  80  to the buffer circuit  35  (FIG. 9) being placed outside. The synthetic filter circuit  88  decodes an LP coefficient portion using information about a packet fed from the input terminal  50 . Then, the synthetic filter circuit  88  constructs a synthetic filter by using the decoded LP coefficient and decodes speech by driving this filter using an exciting signal fed from the adder  75  and passes it to an output terminal  90 . Providing that the LP coefficient is a(i), i=1, . . . , p, decoded speech x(t) can be calculated from an exciting signal e(t) by a following equation:  
               x        (   t   )       =       e        (   t   )       +       ∑     i   =   1     p            a        (   i   )       ×     (     t   -   i     )                   Equation                   (   1   )                                 
 
           [0013]    To solve the equation (1), decoded speech x (t−i), i=1, i=1, . . . , p received in the past is stored as an internal signal through the input/output terminal  80  in the buffer circuit  35  placed outside and is read into the decoding circuit  203  through the input/output terminal  80  when necessary. Here, “p” is an order of the LP coefficient. If the loss information fed from the input terminal  55  indicates occurrence of loss of a packet, the LP coefficient portion decoded from, for example, a previous packet is again used. The input/output terminal  80  outputs an exciting signal fed from the adder  75  as an internal signal to the buffer circuit  35  placed outside. Also, the input/output terminal  80  passes an adaptive code vector fed from the buffer circuit  35  placed outside in accordance with a pitch period fed from the pitch predicting circuit  68  as an internal signal to the pitch predicting circuit  68 . Moreover, the input/output terminal  80  outputs decoded speech received in the past and fed from the synthetic filter circuit  88  as an internal signal to the buffer circuit and receives the decoded speech at a time when a subsequent packet is decoded and passes it to the synthetic filter circuit  88 . The output terminal  90  outputs decoded speech fed from the synthetic filter circuit  88 . In the CELP method, by performing filtering used to accentuate a spectral peak, which is called “post-filtering”, on decoded speech output from the output terminal  90 , acoustic quality of decoded speech can be improved.  
           [0014]    [0014]FIG. 11 is a block diagram showing an example of a decoding circuit employed in a decoding device using the CELP method, in which the decoding circuit  30  shown in FIG. 9 is provided as a decoding circuit  204  in FIG. 11. The ADPCM method is described in “Overview of the ADPCM Coding Algorithm” (IEEE Proc. Of GLOBECOM&#39; 84, pp. 774-777, 1984) (Reference No.  4 ). In the ADPCM-type encoding device, a predicting signal is subtracted from input speech for every sample and a resulting differential signal is encoded by a non-linear adaptive quantizer. Next, by using an output code obtained by the encoding, adaptation and adaptive reverse quantization processes are performed on a scale factor for quantizing. Reproduced speech is obtained by adding a predicting signal to the quantized differential signal obtained by the adaptive reverse quantization. An adaptive predicting device, by using these quantizied differential signal and reproduced speech, calculates a predicting signal. A decoding device performs a decoding process by calculating a predicting signal by same operations as performed in the encoding device. More particularly, the decoding device, by using a received quantized code, performs adaptation and adaptive reverse quantization of a scale factor for quantizing. Next, the adaptive predicting device, by using these quantized differential signal and reproduced speech, calculates a predicting signal of input speech. Finally, reproduced speech is obtained by adding a predicting signal to the quantized differential signal obtained by the adaptive reverse quantization.  
           [0015]    Next, operations of the decoding circuit 204 (ADPCM-type) are described by referring to FIG. 11. When the ADPCM method in which an output code is obtained for every input speech sample is applied to packet communications, quantized codes are combined, for example, every 10 msec and transmitted as one packet. The input terminal  50  receives a packet and passes it to a reverse quantizing circuit  95  and a scale adaptive circuit  110 . The input terminal  55  receives loss information and passes it to the reverse quantizing circuit  95 , the scale adaptive circuit  110 , a speed controlling circuit  115 , and an adaptive predicting circuit  105 . The reverse quantizing circuit  95  decodes a differential signal dp(k) by using a scale coefficient fed from the scale adaptive circuit  110  and by reverse-quantizing a code contained in a packet fed from the input terminal  50  and passes it to an adder  100  and the adaptive predicting circuit  105 . If the loss information fed from the input terminal  55  indicates occurrence of loss of a packet, a signal made up of “zero” is output. The scale adaptive circuit  110  calculates a scale coefficient by using information I(k) contained in a packet fed from the input terminal  50  and a speed controlling coefficient al(k) fed from the speed controlling circuit  115  and passes a result from the calculation to the reverse quantizing circuit  95  and the speed controlling circuit  115 . A scale controlling factor y(k) at a time “k” is obtained using a speed controlling coefficient al(k), a high-speed scale coefficient yu(k−1) received in the past, and a low-speed scale coefficient yl(k−1) by a following equation:  
             y ( k )= al ( k )  yu (k−1)+(1− al ( k ))  yl ( k− 1)  Equation (2)  
           [0016]    Here, a high-speed scale coefficient yu(k) and a low-speed scale coefficient yl(k) at a time “k” are updated, based on the scale controlling coefficient y(k) at the time “k” when the above scale coefficients were calculated, by following equations:  
             yu ( k )=(1−2 −5 ) y ( k )+2 −5 W[ I ( k )]  Equation (3)  
             yl ( k )=(1−2 −6 )  yl ( k −1)+2 −6   yu ( k )  Equation (4)  
           [0017]    where W[X] is a function using “X” as an argument, and reference is made to a predetermined table. Moreover, the scale adaptive circuit  110  outputs a high-speed scale coefficient yu(k) and a low-speed scale coefficient yl (k) both being obtained by solving the equations (3) and (4), as an internal signal from the input/output terminal  80 , stores them in the buffer circuit  35  being placed outside, and then again receives them as a previous sample&#39;s coefficients yu (k−1) and yl (k−1) from the input/output terminal  80  for use when solving the equations (3) and (4) next. When the loss information fed from the input terminal  55  indicates occurrence of loss of a packet, while a concealment process is being performed on the packet, equations (3) and (4) are not updated. The speed controlling circuit  115 , by using following equations, calculates a speed controlling coefficient al (k) from a scale coefficient y(k) fed from the scale adaptive circuit  110 .  
               al        (   k   )       =     {                      1   ,                                   ap        (     k   -   1     )       &gt;   1                            ap        (     k   -   1     )       ,                        ap        (     k   -   1     )       ≤   1                  
        where               Equation                   (   5   )                   ap        (   k   )       =     {                          [     1   -     2     -   4         ]          ap        (     k   -   1     )         +     2     -   3         ,                               dms        (   k   )       -     dml        (   k   )              &gt;       2     -   3            dml        (   k   )                     or                   y        (   k   )         &lt;   3                              [     1   -     2     -   4         ]          ap        (     k   -   1     )         ,                    other                     Equation                   (   6   )                                 
 
             dms ( k )=[1−2 −5   ]dms ( k− 1)+2 −5   F[I ( k )]  Equation (7)  
             dml ( k )=[1−2 −7   ]dml (k−1)+2 −7   F[I ( k )]  Equation (8)  
           [0018]    where F[X] is a function using “X” as an argument, and reference is made to a predetermined table. Moreover, the speed controlling circuit  115  outputs the coefficients ap(k), dms (k), and dml(k) all being obtained by solving the equations (6) to (8) as internal signals from the input/output terminal  80 , stores them in the buffer circuit  35  being placed outside, and then again receives them as a previous sample&#39;s coefficients ap(k−1), dms(k−1) and dml(k−1) from the input/output terminal  80  for use when solving the equations (6) to (8) next. When the loss information fed from the input terminal  55  indicates occurrence of loss of a packet, while a concealment process is being performed on the packet, equations (6) to (8) are not updated. The adaptive predicting circuit  105 , by using a differential signal dp(k) fed from the reverse quantizing circuit  95 , a predicting signal se (k−1), i=1, . . . , 2 received in the past fed through the input/output terminal  80  from the buffer circuit  35  placed outside, and a differential signal dp(k−1), i=1, . . . , 6 received in the past, calculates a predicting signal se(k) at a time “k” by following equations and passes a result from the calculation to the adder  100 .  
               se        (   k   )       =         ∑     i   =   1     2            a        (     i   ,     k   -   1       )            sr        (     k   -   i     )           +     sez        (   k   )                 Equation                   (   9   )                                 
 
           [0019]    where,  
             sr ( k−i )= se ( k−i )+ dq ( k−i )  Equation (10)                sez        (   k   )       =       ∑     i   =   1     6            b        (     i   ,     k   -   1       )            dq        (     k   -   i     )                   Equation                   (   11   )                                   
           [0020]    Moreover, “a(i, k−1)” and “b(i, k−1)” are predicting coefficients and are updated based on dp(k) by following equations so as to be a(i, k) and b(i, k) respectively.  
             b ( i,k )=[1−2 −8   ]b ( i,k− 1)+2 −8   sgn[dq ( k )] sgn[dq ( k−i )], i=1, . . . , 6  Equation (12)  
             a (1 ,k )=[1−2 −8   ]a (1 ,k− 1   )+3·2 −8   sgn[p ( k )] sgn[p ( k− 1   )]  Equation (13)  
             a (2 ,k )=[1−2 −7   ]a (2 ,k− 1)+2 −7   sgn[p ( k )] sgn[p ( k− 2)]− f[a (1 , k− 1)] sgn[p ( k )] sgn[p ( k− 1)]  Equation (14)  
           [0021]    where,  
             p ( k )= dq ( k )+ sez ( k )  Equation (15)                f        (   x   )       =     {             4      x     ,                x        ≤     2     -   1                     2        sgn        (   x   )         ,                x        &gt;     2     -   1                         Equation                   (   16   )                                   
           [0022]    however;  
           | a (2 ,k )|≦0.75  Equation (17)  
           | a (1 ,k )|≦1−2 −4   −a (2 ,k )  Equation (18)  
           [0023]    where sgn [X] represents a code of “x”. The adaptive predicting circuit  105  stores dq(k) fed from the reverse quantizing circuit  95 , se (k) calculated by the equations (9) to (10) and a(i, k) and b(i, k) calculated by the equations (12) to (14) through the input/output terminal  80  in the buffer circuit  35  being placed outside and uses them as a previous sample&#39;s coefficients dp(k−1), se (k−1), a (i, k−1), and b (i, k−1) when solving the equations (9) to (14) next. When the loss information fed from the input terminal  55  indicates occurrence of loss of a packet, while a concealment process is being performed on the packet, equations (12) and (14) are not updated. The adder  100  passes decoded speech obtained by adding up a reverse quantized signal fed from the reverse quantizing circuit  95  and a predicting signal fed from the adaptive predicting circuit  105  to the adaptive predicting circuit  105  and the output terminal  90 . The output terminal  90  outputs the decoded speech fed from the adder  100 . Moreover, in the concealment processing performed according to the ADPCM method, instead of a code I(K) lost due to loss of a packet, a code which makes a reverse quantized signal become zero or a small value (for example, an absolute value is less than  7 ) may be used. This causes decoded speech to become a small value.  
           [0024]    [0024]FIG. 12 is a schematic block diagram showing an example of configurations of the decoding circuit  30  in a band-splitting speech decoding device. When a signal in each band is encoded, various methods including the CELP, the ADPCM method, or a like can be applied. A typical method is an ITU-T G.722 method, which is described in, for example, “7 kHz Audio Coding within 64 kbit/s” (ITU-T Recommendation G. 722, 1988) (Reference No.  5 ).  
           [0025]    Next, operations of the band-splitting type speech decoding circuit are described by referring to FIG. 12. An input terminal  121  receives a packet and passes it to a low-band decoding circuit  66  and a high-band decoding circuit  67 . An input terminal  56  receives loss information and passes it to the low-band decoding circuit  66  and the high-band decoding circuit  67 . The CELP method shown in FIG. 10 and the ADPCM method shown in FIG. 11 can be applied to the low-band decoding circuit  66  and/or the high-band decoding circuit  67 . The low-band decoding circuit  66  decodes speech having signals in a low frequency band (for example, less than 4 kHz) according to the loss information fed from the input terminal  56  by using a packet fed from the input terminal  121  and passes the decoded speech to a band adder  43 . The low-band decoding circuit  66  receives and transmits an internal signal through the input/output terminal  80  from and to the buffer circuit  35  being placed outside. The high-band decoding circuit  67  decodes speech having a band signal corresponding to a high frequency band (for example, 4 kHz or more) according to the loss information fed from the input terminal  56  by using a packet fed from the input terminal  121  and passes the decoded speech to the band adder  43 . Moreover, the high-band decoding circuit  67  receives and transmits an internal signal through the input/output terminal  80  from and to the buffer circuit  35  placed outside. The band adder  43  performs up-sampling on the high-band speech as a component of a high frequency band fed from the high-band decoding circuit  67  and adds this up-sampled speech to a signal obtained by performing up-sampling on the low-band speech as a component of a low frequency band fed from the low-band decoding circuit  66  to decode wide-band speech and passes the decoded speech to an output terminal  51 . The output terminal  51  outputs the wide-band decoded speech fed from the band adder  43 .  
           [0026]    Thus, in the conventional speech decoding device, when loss of a packet occurs, speech corresponding to a portion of speech that has been lost is decoded by using concealment processing. However, the conventional speech decoding device has a problem in that, in the prediction encoding method in which encoding and decoding are performed by using internal signals received in the past, an abnormal large amplitude occurs at a time of decoding packets following the concealment processing and therefore degradation of speech quality occurs. This is because internal signals having not been updated or having been initialized are used in decoding processes, which causes a great difference in internal signals that should be matched between in encoding and decoding processes.  
         SUMMARY OF THE INVENTION  
         [0027]    In view of the above, it is an object of the present invention to provide a speech decoding device and a method for decoding speech being capable of reducing degradation of speech quality caused by concealment processing to be performed when a loss of a packet has occurred.  
           [0028]    According to a first aspect of the present invention, there is provided a speech decoding device including:  
           [0029]    a first circuit to receive a packet and decode speech from the received packet;  
           [0030]    a second circuit to store an internal signal produced in the decoding process by the first circuit and to be used by the first circuit in a decoding process for a subsequent packet to be subsequently received;  
           [0031]    a third circuit to produce concealed speech corresponding to a packet having not been received using a prior received packet; and  
           [0032]    a fourth circuit to update the internal signal using the concealed speech.  
           [0033]    In the foregoing first aspect, a preferable mode is one wherein a code excited linear prediction method is employed and wherein the internal signal contains exciting signals stored as an adaptive code book and prior decoded speech which is to be used in processing by a linear predicting synthetic filter.  
           [0034]    Another preferable mode is one wherein an adaptive differential pulse code modulation method is employed and wherein the internal signal contains a prior output signal which is to be used in predictive processing and coefficients used to control an amplitude or a speed of changing.  
           [0035]    According to a second aspect of the present invention, there is provided a speech decoding device including:  
           [0036]    a decoding circuit to sequentially receive packets containing at least one piece of speech frame data encoded in a block unit for every specified interval in a speech encoding device on a side of a sender, to decode speech frame data in order of packets specified by a time stamp being attached to a received packet, to store an internal signal produced in the decoding process and to be used in a subsequent decoding process for subsequent speech frame data in a buffer, and to produce and output concealed speech corresponding to a packet having not been received, based on the internal signal being stored in the buffer; and  
           [0037]    an updating circuit to update the internal signal being stored in the buffer using an internal signal obtained by encoding the concealed speech produced in the decoding circuit by a same method employed in the speech encoding device.  
           [0038]    In the foregoing second aspect, a preferable mode is one wherein a code excited linear prediction method is employed and wherein the internal signal contains exciting signals stored as an adaptive code book and prior decoded speech which is to be used in processing by a linear predicting synthetic filter.  
           [0039]    Another preferable mode is one wherein an adaptive differential pulse code modulation method is employed and wherein the internal signal contains a prior output signal which is to be used in predictive processing and coefficients used to control an amplitude or a speed of changing.  
           [0040]    According to a third aspect of the present invention, there is provided a speech decoding device including:  
           [0041]    a first circuit to receive a packet and decode speech from the received packet;  
           [0042]    a second circuit to store an internal signal produced in the decoding process by the first circuit and to be used by the first circuit in a decoding process for a subsequent packet to be subsequently received;  
           [0043]    a third circuit to produce concealed speech corresponding to a packet having not been received by using a prior received packet;  
           [0044]    a fourth circuit to measure a length of time during which no receiving of a packet occurs continuously; and  
           [0045]    a fifth circuit to change the internal signal, when the length of time is longer than a predetermined length of time, to decode speech from a packet received thereafter.  
           [0046]    In the foregoing third aspect, a preferable mode is one wherein packets received continuously only within a length of time being shorter than the predetermined length of time are regarded as having not been received in a process of measuring the length of time.  
           [0047]    Another preferable mode is one wherein a code excited linear prediction method is employed and wherein the internal signal contains exciting signals stored as an adaptive code book and prior decoded speech which is to be used in processing by a linear predicting synthetic filter and wherein, in a process of changing the internal signal, a prior signal to be used in predictive processing is made smaller to flatten its spectrum characteristics.  
           [0048]    Still another preferable mode is one wherein an adaptive differential pulse Code modulation method is employed and wherein the internal signal contains a prior output signal which is to be used in predictive processing and coefficients used to control an amplitude or a speed of changing and wherein, in a process of changing the internal signal, a prior signal to be used in predictive processing is made smaller to reduce a prior influence exerted on an amplitude or a change of speed.  
           [0049]    According to a fourth aspect of the present invention, there is provided a speech decoding device including:  
           [0050]    a decoding circuit to sequentially receive packets containing at least one piece of speech frame data encoded in a block unit for every specified interval in a speech encoding device on a side of a sender, to decode speech frame data in order of packets specified by a time stamp attached to a received packet, to store an internal signal produced in the decoding process and to be used in a subsequent decoding process for subsequent speech frame data in a buffer, and to produce and output concealed speech corresponding to a packet having not been received, based on the internal signal being stored in the buffer;  
           [0051]    a loss measuring circuit to measure a length of time during which no receiving of a packet occurs continuously; and  
           [0052]    wherein the decoding circuit is so configured, when the length of time measured by the loss measuring circuit is longer than a predetermined length of time, as to change the internal signal being stored in the buffer for use, to decode speech from a packet received thereafter.  
           [0053]    In the foregoing fourth aspect, a preferable mode is one wherein packets received continuously only within a length of time being shorter than the predetermined length of time are regarded as having not been received in a process of measuring the length of time.  
           [0054]    Another preferable mode is one wherein a code excited linear prediction method is employed and wherein the internal signal contains exciting signals stored as an adaptive code book and prior decoded speech which is to be used in processing by a linear predicting synthetic filter and wherein, in a process of changing the internal signal, a prior signal to be used in predictive processing is made smaller to flatten its spectrum characteristics.  
           [0055]    Still another preferable mode is one wherein an adaptive differential pulse Code modulation method is employed and wherein the internal signal contains a prior output signal which is to be used in predictive processing and coefficients used to control an amplitude or a speed of changing and wherein, in a process of changing the internal signal, a prior signal to be used in predictive processing is made smaller to reduce a prior influence exerted on an amplitude or a change of speed.  
           [0056]    According to a fifth aspect of the present invention, there is provided a method for decoding speech including:  
           [0057]    a first step of receiving a packet and decoding speech from the received packet;  
           [0058]    a second step of storing an internal signal produced by decoding in the first step and to be used in the first step for decoding of a subsequent packet to be subsequently received;  
           [0059]    a third step of producing concealed speech corresponding to a packet having not been received using a prior received packet; and  
           [0060]    a fourth step of updating the internal signal by using the concealed speech.  
           [0061]    In the foregoing fifth aspect, a preferable mode is one wherein a code excited linear prediction method is employed and wherein the internal signal contains exciting signals stored as an adaptive code book and prior decoded speech which is to be used in processing by a linear predicting synthetic filter.  
           [0062]    Another preferable mode is one wherein an adaptive differential pulse code modulation method is employed and wherein the internal signal contains a prior output signal which is to be used in predictive processing and coefficients used to control an amplitude or a speed of changing.  
           [0063]    According to a sixth aspect of the present invention, there is provided a method for decoding speech including:  
           [0064]    a first step of receiving a packet and decoding speech from the received packet;  
           [0065]    a second step of storing an internal signal produced by decoding in the first step and to be used in the first step for decoding of a subsequent packet to be subsequently received;  
           [0066]    a third step of producing concealed speech corresponding to a packet having not been received using a prior received packet;  
           [0067]    a fourth step of measuring a length of time during which no receiving of a packet occurs continuously; and  
           [0068]    a fifth step of changing the internal signal, when the length of time is longer than a predetermined length of time, to decode speech from a packet received thereafter.  
           [0069]    In the foregoing sixth aspect, a preferable mode is one wherein, in the fourth step, packets received continuously only within a length of time being shorter than a predetermined length of time are regarded as having not been received in a process of measuring the length of time.  
           [0070]    Another preferable mode is one wherein a code excited linear prediction method is employed and wherein the internal signal contains exciting signals stored as an adaptive code book and prior decoded speech which is to be used in processing by a linear predicting synthetic filter and wherein, in a process of changing the internal signal, a prior signal to be used in predictive processing is made smaller and a spectrum characteristic is made flattened.  
           [0071]    Still another preferable mode is one wherein an adaptive differential pulse Code modulation method is employed and wherein the internal signal contains a prior output signal which is to be used in predictive processing and coefficients used to control an amplitude or a speed of changing and wherein, in a process of changing the internal signal, a prior signal to be used in predictive processing is made smaller to reduce a prior influence exerted on an amplitude or a speed of changing.  
           [0072]    With the above configurations, by employing an approximation method in which decoded speech produced by concealment processing does not differ greatly from encoded input speech and by encoding the decoded speech produced by concealment processing in a decoding device, internal signals required in the decoding device are updated. The decoded internal signals are used in decoding of a subsequent packet. This enables reduction of mismatching that occurs due to concealment processing between internal signals in the encoding device and internal signals in the decoding device. As a result, quality of decoded speech can be improved. Moreover, if loss of a packet occurs during a long length of time, internal signals in the decoding device become different greatly from internal signals in the coding device. To reduce this difference, in the case of occurrence of loss of a packet during a long length of time, limitation is imposed on internal signals so that first decoded speech on which decoding from a packet is performed does not take on a large value. This also enables reduction of mismatching that occurs due to concealment processing between internal signals in the encoding device and internal signals in the decoding device. As a result, quality of decoded speech can be improved. That is, occurrence of an abnormally large amplitude, that was found in the conventional decoding device, caused by decoding of a packet following concealment processing performed due to loss of a packet can be reduced and degradation in speech quality can be prevented. This is because differences in internal signals occurring between encoding processing and decoding processing can be reduced by updating internal signals using concealed speech by processing being approximate to encoding processing and imposing a limitation on internal signals so that first decoded speech on which decoding from a packet is performed does not take on a large value. 
       
    
    
     BRIEF DESCRIPTION OF THE DRAWINGS  
       [0073]    The above and other objects, advantages, and features of the present invention will be more apparent from the following description taken in conjunction with the accompanying drawings in which:  
         [0074]    [0074]FIG. 1 is a schematic block diagram showing an example of configurations of a speech decoding device according to a first embodiment of the present invention;  
         [0075]    [0075]FIG. 2 is a schematic block diagram showing an example of configurations of an updating circuit employed in the speech decoding device of the first embodiment to which a CELP method is applied;  
         [0076]    [0076]FIG. 3 is a schematic block diagram showing an example of configurations of an updating circuit employed in the speech decoding device of the first embodiment to which an ADPCM method is applied;  
         [0077]    [0077]FIG. 4 is a schematic block diagram showing an example of configurations of an updating circuit employed in the speech decoding device of the first embodiment to which a band-splitting method is applied;  
         [0078]    [0078]FIG. 5 is a schematic block diagram showing an example of configurations of a speech decoding device according to a second embodiment of the present invention;  
         [0079]    [0079]FIG. 6 is a diagram showing an example of configurations of a decoding circuit employed in the speech decoding device of the second embodiment to which a CELP method is applied;  
         [0080]    [0080]FIG. 7 is a schematic block diagram showing an example of configurations of a decoding circuit employed in the speech decoding device of the second embodiment to which an ADPCM method is applied;  
         [0081]    [0081]FIG. 8 is a schematic block diagram showing an example of configurations of a decoding circuit employed in the speech decoding device of the second embodiment to which a band-splitting method is applied;  
         [0082]    [0082]FIG. 9 is a schematic block diagram showing an example of configurations of a speech decoding device based on a conventional speech decoding method;  
         [0083]    [0083]FIG. 10 is a schematic block diagram showing an example of configurations of a speech decoding circuit employed in a conventional speech decoding device to which a CELP method is applied;  
         [0084]    [0084]FIG. 11 is a schematic block diagram showing an example of configurations of a speech decoding circuit employed in the conventional speech decoding device to which an ADPCM method is applied; and  
         [0085]    [0085]FIG. 12 is a schematic block diagram showing an example of configurations of a speech decoding circuit employed in the conventional speech decoding device to which a band splitting method is applied.  
     
    
     DETAILED DESCRIPTION OF THE PREFERRED EMBODIMENTS  
       [0086]    Best modes of carrying out the present invention will be described in further detail using various embodiments with reference to the accompanying drawings.  
       First Embodiment  
       [0087]    A speech decoding device of a first embodiment of the present invention is described by referring to FIG. 1 to FIG. 4. FIG. 1 is a schematic block diagram showing an example of configurations of the speech decoding device according to the first embodiment of the present invention. FIG. 2 is a schematic block diagram showing an example of configurations of an updating circuit  91  employed in the speech decoding device of the first embodiment to which a CELP method is applied. FIG. 3 is a schematic block diagram showing an example of configurations of an updating circuit  92  employed in the speech decoding device of the first embodiment to which an ADPCM method is applied. FIG. 4 is a schematic block diagram showing an example of configurations of an updating circuit  93  employed in the speech decoding device of the first embodiment to which a band-splitting method is applied in which signals in all bands are produced from signals decoded after splitting of a band.  
         [0088]    Configurations of the speech decoding device of the first embodiment shown in FIG. 1 differ from those of the conventional speech decoding device shown in FIG. 9 in that, instead of a buffer circuit  35 , an updating buffer circuit  38  and an updating circuit  40  are newly provided. Only operations related to the updating buffer circuit  38  and the updating circuit  40  are explained accordingly. An input terminal  10  feeds loss information not only to a decoding circuit  30  but also to the updating circuit  40  and the updating buffer circuit  38 . The decoding circuit  30  receives and transmits internal signals from and to the updating buffer circuit  38 . Moreover, the decoding circuit  30  passes decoded speech to the updating circuit  40 . The updating circuit  40 , if the loss information fed from the input terminal  10  indicates occurrence of loss of a packet, by using the decoded speech fed from the decoding circuit  30 , updates internal signals fed from the updating buffer circuit  38  and returns the updated internal signal to the updating buffer circuit  38 . The updating buffer circuit  38 , if the loss information fed from the input terminal  10  indicates occurrence of loss of a packet, receives the updated internal signals from the updating circuit  40  and replaces them with internal signals being stored to be used in processing in the decoding circuit  30 . To simplify the processing, when packets are lost continuously, the above replacement may be performed not on each of lost packets but only on a last one of packets that are lost continuously.  
         [0089]    Operations of the updating circuit  40  to which the CELP method is applied are described by referring to FIG. 2 in which the updating circuit  40  shown in FIG. 1 is shown as an updating circuit  91  in FIG. 2. In the updating circuit  91 , same processing as the encoding according to the CELP method is performed. Details of the encoding processing according to the CELP method are described in, for example, Reference No. 3 . (See Description of the Related Art.) An input terminal  51  receives decoded speech and feeds it to an influence signal subtracting circuit  72  and an LP (Linear Predicting) circuit  71 . An input terminal  56  receives loss information and, only when the loss information indicates occurrence of loss of a packet, performs processing contained in the updating circuit  91 . The influence signal subtracting circuit  72  subtracts influence signal, which was received in the past fed from a synthetic filter circuit  85 , from decoded speech fed from the input terminal  51  and feeds subtracted decoded speech as a result of the substraction to a speech source analyzing circuit  65  and a pitch analyzing circuit  70 . The LP circuit  71  performs an LP (Linear Prediction) analysis on decoded speech fed from the input terminal  51  and performs encoding and decoding of an LP (Linear Prediction) coefficient obtained from the above analysis. Moreover, the LP circuit  71  passes the quantized LP coefficient obtained from decoding to the speech source analyzing circuit  65 , a pitch analyzing circuit  70 , and a synthetic filter circuit  85 . The speech source analyzing circuit  65 , by using the subtracted decoded speech fed from the influence signal subtracting circuit  72  and a quantized LP coefficient fed from the LP circuit  71 , encodes a speech source signal contained in the subtracted decoded speech. Moreover, the speech source analyzing circuit  65  passes the speech source signal to an adder  75  and the pitch analyzing circuit  70 . The pitch analyzing circuit  70 , by using the subtracted decoded speech fed from the influence signal subtracting circuit  72  and the quantized LP coefficient fed from the LP circuit  71 , and an exciting signal obtained from the updating buffer circuit  38  being placed outside through an input/output terminal  121 , extracts a pitch period from the subtracted decoded speech and calculates a corresponding pitch signal. The adder  75  produces an exciting signal by adding up a source signal fed from the speech source analyzing circuit  65  and a pitch period signal fed from the pitch analyzing circuit  70 . Moreover, the adder  75  passes the exciting signal to the synthetic filter circuit  85  and, at a same time, through the input/output terminal  121  to the updating buffer circuit  38  placed outside as an internal signal. The synthetic filter circuit  85  makes up a synthetic filter using the quantized LP coefficient fed from the LP circuit  71  and calculates an influence signal by driving the synthetic filter using the exciting signal fed from the adder  75  and passes the influence signal to the influence signal subtracting circuit  72 . Also, the synthetic filter circuit  85  receives and transmits the influence signal received in the past and to be used in filtering processing through the input/output terminal  121  from and to the updating buffer circuit  38  being placed outside. The input/output terminal  121  is used, in order to output an exciting signal from the adder  75 , to receive and transmit an internal signal used by the synthetic filter circuit  85  and pitch analyzing circuit  70  to and from the updating buffer circuit  38  being placed outside.  
         [0090]    Operations of the updating circuit  40  to which the ADPCM method is applied are described by referring to FIG. 3 in which the updating circuit  40  shown in FIG. 1 is shown as an updating circuit  92 . In the updating circuit  92 , same processing as the encoding according to the ADPCM method is performed. Details of the encoding processing according to the ADPCM method are described in, for example, Reference No. 4 . (See Description of the Related Art.) The input terminal  51  receives decoded speech and passes it to a differential circuit  76 . The differential circuit  76  subtracts a predicting signal fed from an adaptive predicting circuit  105  from the decoded speech fed from the input terminal  51  and passes the obtained differential signal to a quantizing circuit  25 . The quantizing circuit  25  scalar-quantizes the differential signal fed from the differential circuit  76  and passes obtained quantized codes to a reverse quantizing circuit  95  and a scale adaptive circuit  110 . The reverse quantizing circuit  95 , by using a scale coefficient fed from the scale adaptive circuit  110 , decodes the quantized differential signal from the quantized codes fed from the quantizing circuit  25  by using reverse quantizing processing and outputs them to an adder  100  and the adaptive predicting circuit  105 . The scale adaptive circuit  110 , by using the quantized codes fed from the quantizing circuit  25  and a speed controlling coefficient fed from a speed controlling circuit  115 , calculates a scale coefficient and passes it to the reverse quantizing circuit  95  and the speed controlling circuit  115 . A scale coefficient y(k) is calculated by the equations (2) to (4) described above using a speed controlling coefficient al(k), a high-speed scale coefficient yu(k), and a low-speed coefficient yl (k). Moreover, the scale adaptive circuit  110  outputs the high-speed scale coefficient yu(k) and low-speed coefficient yl (k) calculated by the equations (3) and (4) (Description of the Related Art) from the input/output terminal  121 , then stores them in the updating buffer circuit  38  being placed outside and again receives them from the input/output terminal  121  as a previous sample&#39;s coefficients yu(k−1) and yl(k−1) for use when solving the equations (3) and (4) next. The speed controlling circuit  115 , by using the equations (5) to (8) described above, calculates a speed controlling coefficient al(k) from the scale coefficient y(k) fed from the scale adaptive circuit  110 . Also, the speed controlling circuit  115  outputs the coefficients ap(k), dms(k), and dml(k) calculated by the equations ( 6 ) to ( 8 ) (Description of the Related Art) from the input/output terminal  121 , passes them to the updating buffer circuit  38  being placed outside, then again receives them, from the input/output terminal  121 , as a previous sample&#39;s coefficients ap(k−1), dms(k−1), and dml(k−1) for use when solving the equations (6) to (8) next. The adaptive predicting circuit  105 , by using the differential signal dq(k) fed from the reverse quantizing circuit  95 , the predicting signal se (k-i), i=1, . . . , 2 received in the past and fed from the input/output terminal  121 , and the differential signal dq(k−i), i=1, . . . , 6 received in the past, calculates a predicting signal at a time “k” by the equations (9) to (11) (See Description of the Related Art) described above and passes it to the adder  100 . Here, the coefficients a(i, k−1) and b(i, k−1) are predicting coefficients and are updated to be coefficients a(i, k) and b(i, k) based on the differential signal dq(k) (refer to the equations (See Description of the Related Art) (12) to (14)). Also, the adaptive predicting circuit  105  feeds dq(k) fed from the reverse quantizing circuit  95 , se(k) calculated by the equations (9) to (11), and a(i, k) and b(i, k) calculated by the equations (12) to (14) through the input and output terminal  121  to the updating buffer circuit  38  being placed outside and uses them as a previous sample&#39;s values dq(k−1), se(k−1), a(i, k−1), and b(i, k−1) when solving the equations (9) to (14) next. The adder  100  passes decoded speech obtained by adding up the reverse quantized signal fed from the reverse quantizing circuit  95  and the predicting signal fed from the adaptive predicting circuit  105  to the adaptive predicting circuit  105  and the output terminal  45 .  
         [0091]    Operations of the updating circuit to which the band-splitting method is applied are described by referring to FIG. 4 in which the updating circuit  40  shown in FIG. 1 is shown as an updating circuit  93 . The updating circuit  93  performs same processing as a band-splitting encoding method designated by ITU-T G.722 or a like and details of the method are described in, for example, Reference No. 5 . (See Description of the Related Art) The input terminal  51  receives the decoded speech and passes it to a band-splitting circuit  43 . The input terminal  56  receives loss information and, only if the loss information indicates occurrence of loss of a packet, performs processing contained in the updating circuit  93 . The band-splitting circuit  43  splits the decoded speech into a high-band signal having a high frequency band component and being down-sampled and into a low-band signal having a low frequency band component. Moreover, the band-splitting circuit  43  passes the high-band signal and the low-band signal, respectively, to a high-band buffer updating circuit  42  and to a low-band buffer updating circuit  41 . As the high-band buffer updating circuit  42  and low-band buffer updating circuit  41 , each of the updating circuits  91  and  92  shown in detail in FIG. 2 and FIG. 3 may be used. The low-band buffer updating circuit  41  encodes a low-band signal fed from the band-splitting circuit  43 . At this time, the low-band buffer updating circuit  41  receives and transmits an internal signal through the input/output terminal  121  from and to the updating buffer circuit  38  being placed outside. The high-band buffer updating circuit  42  encodes a high-band signal fed from the band-splitting circuit  43 . At this time, the high-band buffer updating circuit  42  receives and transmits an internal signal through the input/output terminal  121  from and to the updating buffer circuit  38  being placed outside. Moreover, when a band-splitting method is applied to a speech decoding device, that is, when a decoding circuit shown in FIG. 12 (Prior Art) is used as the decoding circuit  30  shown in FIG. 1 and the updating circuit  93  shown in FIG. 4 is used as the updating circuit  40  shown in FIG. 1, it is not necessary that decoded speech is fed from the decoding circuit  30  shown in FIG. 1 to the updating circuit  40  shown in FIG. 1 and a low-band decoded signal calculated by a low-band decoding circuit  66  shown in FIG. 12 (Prior Art) may be directly passed to the low-band buffer updating circuit  41  shown in FIG. 4 and a high-band decoded signal calculated by a high-band decoding circuit  67  shown in FIG. 12 may be directly passed to the high-band buffer updating circuit  42  shown in FIG. 4. By configuring above, the band-splitting circuit  43  shown in FIG. 4 can be removed and an amount of arithmetic operations can be reduced.  
       Second Embodiment  
       [0092]    A speech decoding device of a second embodiment of the present invention is described by referring to FIG. 5 to FIG. 8. FIG. 5 is a schematic block diagram showing an example of configurations of the speech decoding device according to the second embodiment. FIG. 6 is a decoding circuit  200  employed in the speech decoding device of the second embodiment to which a CELP method is applied. FIG. 7 is a schematic block diagram showing an example of configurations of a decoding circuit  201  employed in the speech decoding device of the second embodiment to which an ADPCM method is applied. FIG. 8 is a schematic block diagram showing an example of configurations of a decoding circuit employed in the speech decoding device of the second embodiment to which a band-splitting method is applied in which signals in all bands are produced from signals decoded after splitting of a band. Configurations of the decoding device of the second embodiment differ from those in the conventional one shown in FIG. 9 only in that a conventional decoding circuit  30  is replaced with a decoding circuit  33 , and a loss measuring circuit  20  is newly provided only operations related to these components are explained accordingly. An input terminal  10  passes loss information not only to the decoding circuit  33  but to the loss measuring circuit  20 . The loss measuring circuit  20 , by using loss information fed from the input terminal  10 , measures a number of times of continuous losses or a length of time of the loss and feeds a result from the measurement to the decoding circuit  33 . The decoding circuit  33 , unlike in the case of the conventional one, by using not only the loss information fed from the input terminal  10  but also the result from the measurement fed from the loss measuring circuit  20 , decodes speech from packets fed from an input terminal  15 . More particularly, the decoding circuit  33 , if time obtained from the above measurement is longer than a predetermined time, changes an internal signal when speech is decoded from packets that arrived thereafter.  
         [0093]    Next, the decoding circuit  33  of the second embodiment is described by referring to FIG. 6 and FIG. 7. First, operations of the decoding circuit  33  performed when the CELP method is employed are described by referring to FIG. 6 in which the decoding circuit  33  shown in FIG. 5 is provided as a decoding circuit  200  in FIG. 6. Configurations of the decoding circuit  200  shown in FIG. 6 differ from those of a conventional CELP-type decoding circuit  203  shown in FIG. 10 in that a speech source analyzing circuit  65 , a pitch predicting circuit  68 , and a synthetic filter circuit  88  are replaced respectively with a speech source circuit  64 , a pitch predicting circuit  69 , and a synthetic filter circuit  85  and there is additionally provided with an input terminal  60  to receive a result from measurement of a number of times of loss. Only operations related to these components are explained accordingly. The input terminal  60  receives a result of the measurement and passes it to the speech source circuit  64 , the pitch predicting circuit  69 , and the synthetic filter circuit  85 . Configurations of the speech source circuit  64  of the embodiment differ from those of the conventional speech source analyzing circuit  65  in that, if time being a result from the above measurement fed from the input terminal  60  exceeds a predetermined number of times of loss or a length of time of loss, a speech signal is produced by attenuating a gain of the speech source code vector. An amount of attenuation should be, for example, about 3 dB so as to avoid discontinuous decoded speech. Moreover, the pitch predicting circuit  69  of the embodiment differ from those of the conventional pitch predicting circuit  68  in that, if the result from the measurement fed from the input terminal  60  exceeds the predetermined number of times of loss or the predetermined length of time of loss, a pitch signal is produced by reducing a gain of an adaptive code vector. An amount of attenuation should be, for example, about 3 dB so as to avoid discontinuous decoded speech.  
         [0094]    Configurations of the synthetic filter circuit  85  of the embodiment differ from those of the conventional synthetic filter circuit  88  in that, if a result from the measurement fed from the input terminal  60  exceeds the predetermined number of times or the predetermined length of time, filtering is performed after processing of making a spectrum characteristic more flattened has been performed on an LP coefficient of a synthetic filter. As a method for making a spectrum characteristic flattened, a method is available in which a crest of a spectrum is made lower by multiplying an LP coefficient a(i) by β i . Here, β&lt;1. This processing enables reduction of an unwanted voice such as an oscillation sound produced due to a crest of a spectrum possessed by an LP coefficient received in the past.  
         [0095]    Next, operations of the decoding circuit  33  performed when the ADPCM method is employed are described by referring to FIG. 7 in which the decoding circuit  33  shown in FIG. 7 is provided as a decoding circuit  201 . Configurations of the decoding circuit  201  shown in FIG. 7 differ from those of the conventional ADPCM-type decoding circuit  204  shown in FIG. 11 in that a scale adaptive circuit  110 , a speed controlling circuit  115 , and an adaptive predicting circuit  105  are replaced respectively with a scale adaptive circuit  111 , a speed controlling circuit  116 , and an adaptive predicting circuit  106 , and in that there is additionally provided with an input terminal  60  to receive a result from measurement of a number of times of loss. Only operations related to these components are explained accordingly. The input terminal  60  receives a result of the measurement and passes it to the scale adaptive circuit  111 , the speed controlling circuit  116 , and the adaptive predicting circuit  106 . Configurations of the scale adaptive circuit  111  of the embodiment differ from those of the conventional scale adaptive circuit  110  in that, if a result from the measurement fed from the input terminal  60  exceeds a predetermined number of times of loss or a predetermined length of time of loss, calculations are performed by making a little larger than 2 −5  or 2 −6  of coefficients of a right side of each of the equation (3) and (4) (See Description of the Related Art) described above, during a predetermined time interval (for example, during 5 msec of a head). By making these values larger, an influence on yu(k) and yl (k) incurred by an state existed in the past due to updating of the equations (3) and (4) can be reduced and therefore an influence suffered by loss of a packet can be reduced. By performing this processing during a specified short period of time, the influence suffered by a state existed in the past can be sufficiently reduced. Configurations of the speed controlling circuit  116  of the embodiment differ from those of the conventional speed controlling circuit  115  in that, if a result from the measurement fed from the input terminal  60  exceeds a predetermined number of times of loss or a predetermined length of time of loss, calculations are performed by making a little larger than 2 −5  or 2 −7  of coefficients of a right side of each of the equation (7) and (8) (See Description of the Related Art) described above during a predetermined time interval (for example, during 5 msec of a head). By making these values larger, an influence on dms(k) and dml(k) incurred by an state existed in the past due to updating of the equations (7) and (8) (See Description of the Related Art) can be reduced and therefore an influence suffered by loss of a packet can be reduced. Configurations of the adaptive predicting circuit  106  of the embodiment differ from those of the conventional adaptive predicting circuit  105  in that, if a result from the measurement fed from the input terminal  60  exceeds a predetermined number of times of loss or a predetermined length of time of loss, calculations are performed by making a little larger than 2 −8 , 2 −8  or 2 −7  of coefficients of a right side of each of the equation (12), (13) and (14) (See Description of the Related Art) described above, during a predetermined time interval (for example, during 5 msec of a head). By making these values larger, an influence on b(i, k) and a(i, k) incurred by an state existed in the past due to updating of the equations (12) and (14) can be reduced and therefore an influence suffered by loss of a packet can be reduced. Though the processing of making the coefficients larger is performed in the scale adaptive circuit  111 , the speed controlling circuit  116 , and the adaptive predicting circuit  106 , in order to simplify the processing, only any one of the processing executed in these circuits maybe performed. However, effects that can be obtained by the processing decrease.  
         [0096]    Lastly, operations of the decoding circuit  33  performed when the band-splitting method is employed are described by referring to FIG. 8. Configurations of the decoding circuit of the embodiment differ from those of the conventional band-splitting type decoding circuit shown in FIG. 12 in that a low-band decoding circuit  66  and a high-band decoding circuit  67  are replaced respectively with a low-band decoding circuit  81 , a high-band decoding circuit  82 , and there is additionally provided with the input terminal  60  to receive a result from measurement of a number of times of loss. Only operations related to these components are explained accordingly. The input terminal  60  receives a result from the measurement and passes it to the low-band decoding circuit  81  and the high-band decoding circuit  82 . Configurations of the low-band decoding circuit  81  of the embodiment differ from those of the conventional low-band decoding circuit  66  in that an internal signal is controlled according to a result from the measurement fed from the input terminal  60 . Configurations of the high-band decoding circuit  82  of the embodiment differ from those of the conventional high-band decoding circuit  67  in that an internal signal is controlled according to a result from the measurement fed from the input terminal  60 . Here, as the low-band decoding circuit  81  and the high-band decoding circuit  82 , the decoding circuits described in FIG. 6 or FIG. 7 may be used.  
         [0097]    Moreover, in the speech decoding device of the second embodiment of the present invention, when a length of time during which packets are lost continuously is measured, if a length of time of an interval during which packets are received which exists between two intervals during packets are lost is not greater than a predetermined length of time (for example, 10 msec or a length of time corresponding to one packet), the interval between two intervals during which packets are lost can be regarded as continuous. When packets are lost in a short cycle (for example, every packet), unless each of intervals during which packets are lost in a short cycle is regarded as continuous, and a discontinuous feeling in decoded speech occurs due to changes of interval signals in a short cycle. Therefore, by regarding each of the above intervals as continuous, such the discontinuous feeling in the decoded speech can be prevented.  
         [0098]    It is apparent that the present invention is not limited to the above embodiments but may be changed and modified without departing from the scope and spirit of the invention.