Abstract:
An echo removing apparatus for reducing the echo caused by sound returning from a speaker to a microphone of a small-sized communication terminal, such as a portable telephone. The echo removing apparatus includes a filter unit for outputting a pseudo echo signal estimating an echo component returning to the microphone based upon a generated sound signal supplied to the speaker, a subtractor for subtracting the pseudo echo signal supplied from the filter unit from the sound generated collection signal supplied from the microphone and a first characteristics conversion unit for converting frequency characteristics of sound generated collection signals from the microphone on the frequency axis. The characteristics of the filter unit are controlled for minimizing an error component between an output signal of the first characteristics conversion unit and an output signal of the filter unit.

Description:
BACKGROUND OF THE INVENTION 
     This invention relates to an echo removing apparatus and, more particularly, to an echo removing apparatus for reducing the echo caused by the sound generated returning from a speaker to a microphone of a small-sized communication terminal, such as a portable telephone. 
     In keeping up with reduction in size of the sound generated communication terminal, such as a portable telephone, the effect as of the echo produced by the sound generated returning from the sound generated-receiving speaker to the sound generated-sending microphone becomes hardly negligible. For removing the echo caused by the sound generated returning round on the transmitter/receiver, an echo removing apparatus or an echo canceler shown for example in  FIG. 1  is employed. 
     Referring to  FIG. 1 , a terminal  11  receives a speaker output signal x(k) transmitted from a communication partner to a speaker  12 , where k denotes a sample number or a time position of discrete signals. A microphone input signal y(k), collected by a microphone  13  and thereby converted into an electrical signal, is supplied along with a pseudo echo signal supplied from a filter circuit  15  to a subtractor  14 . The subtractor subtracts the pseudo echo signal supplied from the microphone input signal to form a resultant echo-reduced signal or a residual echo signal e(k) which is supplied to an input terminal  16 . In a portable telephone, the speaker  12  and the microphone  13  are usually arranged close to each other as a telephone handset. 
     For the adaptive filter  15 , a so-called finite response (FIR) filter is employed. The filter coefficients or tap coefficients are set for minimizing the error signal(k). The adaptive filter  15  filters the input signal, that is the speaker output signal x(k), for estimating the echo signal for generating the pseudo echo signal. This pseudo echo signal is provided to the subtractor  14  where the pseudo echo signal is subtracted from the microphone input signal y(k) to derive the residual echo signal or error signal e(k). 
     That is, if the input signal supplied from the terminal  11 , that is the speaker output signal x(k), is the tap input to the N-tap FIR filter, operating as the adaptive filter  15 , and the tap coefficients of the adaptive filter  15  are b k (i), where i=0, 1, . , N−1, the pseudo echo signal outputted by the adaptive filter  15  is given by 
     
       
         
           
             
               
                 
                   
                     
                       y 
                       ^ 
                     
                     ⁡ 
                     
                       ( 
                       k 
                       ) 
                     
                   
                   = 
                   
                     
                       ∑ 
                       
                         i 
                         = 
                         0 
                       
                       
                         N 
                         - 
                         1 
                       
                     
                     ⁢ 
                     
                       
                         
                           b 
                           k 
                         
                         ⁡ 
                         
                           ( 
                           i 
                           ) 
                         
                       
                       ⁢ 
                       
                         x 
                         ⁡ 
                         
                           ( 
                           
                             k 
                             - 
                             1 
                           
                           ) 
                         
                       
                     
                   
                 
               
               
                 
                   ( 
                   1 
                   ) 
                 
               
             
           
         
       
     
     The subtractor  14  subtracts the pseudo echo signal of the equation (1) from the microphone input signal y(k) to derive the residual echo signal or error signal e(k) by:
 
 e ( k )= y ( k )− ŷ ( k )  (2)
 
     The tap coefficient {b k (i)} of the adaptive filter  15 , where i=1, 2, . . . , N−1, is updated, by a suitable algorithm, such as a least mean square (LMS) algorithm, learning identification method or a recursive least square (RLS) algorithm, for minimizing the time average of the power of the error signal e(k) of the equation (2), that is,
 
 E[∥e ( k )∥ 2 ]
 
where E[ ] is an expected value or a mean value of the value within the brackets [ ] and ∥e(k)∥ 2  is the square sum of e(k). The tap coefficients of N taps of the adaptive filter  15  are equivalent to estimated values of the echo characteristics between the speaker  12  and the microphone  13 .
 
     Meanwhile, if desired to reduce the number of tapes N to the smallest possible value for simplifying the structure, the frequency characteristics of the residual echo are left in the low frequency range, even though the echo characteristics can be estimated partially with high accuracy by the adaptive filter  15 , thus raising difficulties in effectively canceling the echo from sound generated input signals containing strong low-frequency components. 
       FIG. 2A  shows the impulse response of the echo signals in association with the respective taps of the FIR filter.  FIG. 2B , on the other hand, shows the impulse response of the residual echo signal obtained on subtracting the pseudo echo signal estimated by 20-tap FIR adaptive filter. It is seen from  FIGS. 2A and 2B  that the impulse response of the residual echo signal corresponding to 20 taps has been removed. 
       FIG. 3  shows, in association with  FIGS. 2A and 2B , a spectral curve  a  of the echo signal and a spectral curve  b  of the residual echo signal produced on subtracting the estimated echo signal supplied from the 20-tap FIR adaptive filter. 
     Thus, even if it may appear that the echo characteristics can be substantially estimated by the 20-tap FIR adaptive filter, the frequency characteristics of the residual echo characteristics (curve  b ) are left in a region from 500 Hz to 1 kHz where the energy of sound generated signal would be concentrated to a larger extent. 
     SUMMARY OF THE INVENTION 
     It is therefore an object of the present invention to provide an echo removing apparatus in which satisfactory echo cancellation characteristics may be achieved even with a small number of taps of the FIR adaptive filter employed for estimating echo signals. 
     According to the present invention, there is provided an echo removing apparatus for removing the echo produced by sound generated from a sound generating means being returned to a sound collecting means arranged in proximity to the sound generating means. The echo removing apparatus includes a filter for outputting a pseudo echo signal estimating an echo component returning to a sound generated collecting means based upon a generated sound signal supplied to the sound generating means, subtraction means for subtracting the pseudo echo signal supplied from said filter means from sound generated collection signal supplied from said sound generated collection means, and first characteristics conversion means for converting frequency characteristics of sound generated collection signal supplied from sound generated collection means on the frequency axis. The characteristic of the filter means are control led for minimizing an error component between an output signal of the first characteristics conversion means and an output signal of the filter means. 
     The subtraction means subtracts the pseudo echo signal supplied from an output signal of the first characteristics conversion means. A second characteristics conversion means is also provided in the echo removing apparatus for converting frequency characteristics of an output signal of the subtraction means on the frequency axis. 
     The second characteristics conversion means is provided in the echo removing apparatus for converting frequency characteristics of an output signal of the filter means on the frequency axis. An output signal of the second characteristics conversion means is provided as a pseudo echo signal to the subtraction means for subtraction from sound generated collection signal supplied from sound generated collection means. 
     By subtracting the pseudo echo signal supplied from the signal corresponding to sound generated collection signal whose characteristics have been converted, and by adaptively controlling the characteristics of the filter means for minimizing the resulting error components, the echo cancellation characteristics may be improved even with a smaller number of taps of the filter means. In addition, a smaller processing volume suffices for achieving the echo cancellation characteristics comparable to those of a conventional echo removing apparatus. 
    
    
     
       BRIEF DESCRIPTION OF THE DRAWINGS 
         FIG. 1  is a schematic block diagram showing the structure of a conventional echo removing apparatus. 
         FIGS. 2A and 2B  are graphs showing the response to the tap numbers of an FIR adaptive filter for echo estimation. 
         FIG. 3  is a graph showing a spectral curve for echo signals and a spectral curve for residual echo signals. 
         FIG. 4  is a schematic block diagram showing a basic structure of an echo removing apparatus according to the present invention. 
         FIG. 5  is a schematic block diagram showing another basic structure of an echo removing apparatus according to the present invention. 
         FIG. 6  is a block diagram showing an FIR adaptive filter employed as a filter for converting characteristics of the echo removing apparatus according to the present invention. 
         FIG. 7  is a graph showing frequency characteristics of a fixed-coefficient FIR filter employed as a filter for converting characteristics of the echo removing apparatus according to the present invention. 
         FIG. 8  is a block diagram showing a fixed-coefficient FIR filter employed as a filter for converting characteristics of the echo removing apparatus according to the present invention. 
         FIG. 9  is a block diagram showing an echo removing apparatus according to the present invention on the decoder side of sound generated encoding system. 
         FIG. 10  is a graph showing the relation between the pitch period of the input signal and the number of filter taps. 
         FIG. 11  is a graph illustrating input signal power fluctuations in case of a smaller number of filter taps. 
         FIG. 12  is a graph showing the amount of echo cancellation and the number of filter taps. 
     
    
    
     DETAILED DESCRIPTION OF THE PREFERRED EMBODIMENTS 
     Referring to the drawings, certain preferred embodiments of the present invention will be explained in detail. 
       FIG. 4  schematically shows an embodiment of an echo removing apparatus according to the present invention. To a terminal  11  is supplied a speaker output signal x(k) as a generated sound signal transmitted from a communication partner to a speaker  12 , as the sound generating means. A microphone input signal y(k), collected by a microphone  13 , arranged as sound collection means close to the speaker  12  as sound generating means, and thereby converted into an electrical signal, is converted by a characteristics conversion filter  21 , as first characteristics conversion means, into a signal u(k), which is supplied to an echo-removing subtractor  14 . The subtractor  14  subtracts the pseudo echo signal supplied from the adaptive filter  15  to form an echo-reduced signal or a residual echo signal e(k). This signal is filtered by a characteristics conversion filter  22  as second characteristics conversion means to form a signal z(k) which is outputted at a terminal  16  as an echo-reduced output signal. 
     In the present echo removing apparatus, employed as an example for the sound generated communication terminal of, for example, a portable telephone, the speaker  12  and the microphone  13  are usually arranged close to each other as a handset of a portable telephone. 
     The adaptive filter  15  may, for example, be a finite impulse response (FIR) filter having its filter coefficients or tap coefficients selected by adaptive processing which will minimize a time average value of the power of the error signal e(k). The adaptive filter  15  receives the input signal supplied from the terminal  11 , which is the speaker output signal x(k), as a tap input, and outputs a pseudo echo signal, which estimates the signal u(k) from the characteristics conversion filter  21 , to the subtractor  14 . 
     The first characteristics conversion filter  21  converts characteristics of the input signal, that is the microphone input signal y(k), on the frequency axis. As an illustrative example, the filter  21  preferably has the characteristics of equalizing or whitening the input sound generated signal on the frequency axis. Although the second characteristic conversion filter  22  may be omitted, the second characteristic conversion filter  22 , if used, preferably has the characteristics of canceling the filtering performed by the first characteristic conversion filter  21 . That is, if the transfer functions of the first characteristic conversion filter  21  and the second characteristic conversion filter  22  are W 1 (z) and W 2 (z), respectively, these transfer functions preferably satisfy the relation:
 
 W   1 ( z )* W   2 ( z )=1
 
     For these filters, digital filters of the first or higher order are employed. 
     Referring to  FIG. 5 , the basic structure slightly different from that shown in  FIG. 4  is now explained. 
     In  FIG. 5 , a speaker output signal x(k) as a generated sound signal transmitted from a communication partner is supplied via a terminal  11  to a speaker  12  as the sound generating means. A microphone input signal y(k), collected by a microphone  13 , arranged as a sound collection means close to the speaker  12  as the sound generating means, and thereby converted into an electrical signal, is supplied to a characteristics conversion filter  31 , as a first characteristics conversion means, and to an echo-removing subtractor  14 . The microphone input signal y(k) is filtered by a characteristics conversion filter  31  into a signal u(k) which is supplied to a subtractor  33  where the adaptive filter output signal supplied from the adaptive filter  15  is subtracted from the signal u(k) to form an error signal e(k). This error signal e(k) is supplied to a characteristic conversion filter  32  as the second characteristic conversion means where it is filtered to form a pseudo echo signal which is provided to the subtractor  14 . The subtractor  14  subtracts the pseudo echo signal supplied from the microphone input signal y(k) to form an echo-reduced output signal z(k) which is outputted at a terminal  16 . In the present echo removing apparatus, employed as an example for a sound generated communication terminal of, for example, a portable telephone, the speaker  12  and the microphone  13  are usually arranged close to each other as a handset of a portable telephone. 
     The adaptive filter  15  may, for example, be a finite impulse response (FIR) filter having its filter coefficients or tap coefficients selected by adaptive processing which will minimize a time average value of the power of the error signal e(k). The adaptive filter  15  receives an input signal at the terminal  11 , which is the speaker output signal x(k), as a tap input, and outputs a pseudo echo signal, which estimates the signal u(k) from the characteristics conversion filter  21 , to a subtractor  30 . 
     The first characteristics conversion filter  31  converts characteristics of the input signal, that is the microphone input signal y(k), on the frequency axis. As an illustrative example, the filter  31  preferably has the characteristics of equalizing or whitening the input sound generated signal on the frequency axis. Although the second characteristic conversion filter  32  may be omitted, the second characteristic conversion filter  32 , if used, preferably has the characteristics of canceling the filtering performed by the first characteristic conversion filter  31 . That is, if the transfer functions of the first characteristic conversion filter  31  and the second characteristic conversion filter  32  are W 1 (z) and W 2 (z) , respectively, these transfer functions preferably satisfy the relation:
 
 W   1 ( z )* W   2 ( z )=1
 
     For these filters, digital filters of the first or higher order are employed. 
     In the basic structure, shown in  FIG. 5 , the processing which is equivalent to the basic structure shown in  FIG. 4  is performed by a signal flow different from that of  FIG. 4 . If the processing is realized with a digital signal processor (DSP), the gain of the transfer function W 1 (z) of the first characteristic conversion filter  31  may conveniently be controlled for effectively scaling the filter coefficients without distorting the sound generated by the speaker and talking at the microphone. 
     Several illustrative examples of the basic structures shown in  FIGS. 4 and 5  will now be explained. Although only the basic structure of  FIG. 4  is explained for simplicity of explanation, it should be noticed that the same holds for the basic structure shown in  FIG. 5  as well. 
       FIG. 6  shows a filter modified from the first characteristics conversion filter  21  of  FIG. 4 , in which filter coefficients of a transfer function W 1 (z) of the first characteristics conversion filter  21  have been adaptively changed so that the filter operates as an inverse filter or a whitening filter with respect to the residual echo signal or as the error signal e(k). The transfer function W 2 (z) of the second characteristic conversion filter is set to 1/W 1 (z). 
     In  FIG. 6 , an FIR filter having N tapes is employed as the adaptive filter  15 . If the tap coefficients are represented as b k (i), where i=0, 1, . . . , N−1, the transfer function B k (k) is represented as 
                       B   k     ⁡     (   z   )       =       ∑     i   =   0       N   -   1       ⁢           ⁢         b   k     ⁡     (   i   )       ⁢     z     -   1                   (   3   )               
It is assumed that the transfer function W 1 (z) of an M-tap adaptive filter  41  equivalent to the first characteristic conversion filter  21  of  FIG. 4  is given by
 
     
       
         
           
             
               
                 
                   
                     
                       W 
                       1 
                     
                     ⁡ 
                     
                       ( 
                       z 
                       ) 
                     
                   
                   = 
                   
                     
                       1 
                       - 
                       
                         
                           A 
                           k 
                         
                         ⁡ 
                         
                           ( 
                           z 
                           ) 
                         
                       
                     
                     = 
                     
                       1 
                       - 
                       
                         
                           ∑ 
                           
                             i 
                             = 
                             1 
                           
                           M 
                         
                         ⁢ 
                         
                           
                             
                               a 
                               k 
                             
                             ⁡ 
                             
                               ( 
                               i 
                               ) 
                             
                           
                           ⁢ 
                           
                             z 
                             
                               - 
                               1 
                             
                           
                         
                       
                     
                   
                 
               
               
                 
                   ( 
                   4 
                   ) 
                 
               
             
           
         
       
     
     In such case, a filter  42  is an infinite impulse filter (IIR) having its transfer function W 2 (z), equivalent to the transfer function W 2 (z) of the second characteristic conversion filter  22  of  FIG. 4 , represented by: 
                       W   2     ⁡     (   z   )       =       1     1   -       A   k     ⁡     (   z   )           =     1     1   -       ∑     i   =   0     M     ⁢         a   k     ⁡     (   i   )       ⁢     z     -   i                         (   5   )               
The filter  42  is otherwise the same in structure to the filter of  FIG. 4  and hence the corresponding portions are designated by the same numerals and the description therefor is omitted for simplicity.
 
     In the embodiment of  FIG. 6 , the filtering by the FIR adaptive filter  41  is performed to derive a signal u(k) shown by the equation (6): 
     
       
         
           
             
               
                 
                   
                     u 
                     ⁡ 
                     
                       ( 
                       k 
                       ) 
                     
                   
                   = 
                   
                     
                       y 
                       ⁡ 
                       
                         ( 
                         k 
                         ) 
                       
                     
                     - 
                     
                       
                         ∑ 
                         
                           i 
                           = 
                           1 
                         
                         M 
                       
                       ⁢ 
                       
                         
                           
                             a 
                             k 
                           
                           ⁡ 
                           
                             ( 
                             i 
                             ) 
                           
                         
                         ⁢ 
                         
                           y 
                           ⁡ 
                           
                             ( 
                             
                               k 
                               - 
                               i 
                             
                             ) 
                           
                         
                       
                     
                   
                 
               
               
                 
                   ( 
                   6 
                   ) 
                 
               
             
           
         
       
     
     On the other hand, the adaptive filter  15  receives the speaker output signal x(k), that is an input signal supplied from the terminal  11  , as a tap input signal, and generates an adaptive filter output signal given by the equation: 
     
       
         
           
             
               
                 
                   
                     
                       u 
                       ^ 
                     
                     ⁡ 
                     
                       ( 
                       k 
                       ) 
                     
                   
                   = 
                   
                     
                       ∑ 
                       
                         i 
                         = 
                         0 
                       
                       
                         N 
                         - 
                         1 
                       
                     
                     ⁢ 
                     
                       
                         
                           b 
                           k 
                         
                         ⁡ 
                         
                           ( 
                           i 
                           ) 
                         
                       
                       ⁢ 
                       
                         x 
                         ⁡ 
                         
                           ( 
                           
                             k 
                             - 
                             i 
                           
                           ) 
                         
                       
                     
                   
                 
               
               
                 
                   ( 
                   7 
                   ) 
                 
               
             
           
         
       
     
     The subtractor  14  subtracts an adaptive filter output from the adaptive filter  15  as shown by the equation (7) from the above signal U(k) from the FIR adaptive filter  41  to give the error signal e(k) shown by the equation (8):
 
 e   9   k )= u ( k )− û ( k )  (8)
 
     The error signal e(k) from the subtractor  14  is filtered by the IIR filter  42  to give an echo-reduced output signal z(k) given by the equation: 
                     z   ⁡     (   k   )       =       e   ⁡     (   k   )       +       ∑     i   =   1     M     ⁢         a   k     ⁡     (   i   )       ⁢     e   ⁡     (     k   -   i     )                     (   9   )               
which signal z(k) is outputted at the terminal  16 .
 
     In the above adaptive filtering, the tap coefficients {b k (i)} of the adaptive filter  15 , where i=0, 1, . . . , N−1, and the tap coefficients {a k (i)} of the FIR adaptive filter  41 , where i=0, 1, . . . , M, are updated to {b k+1 (i)} and {a k+1 (i)}, respectively, using a suitable adaptive algorithm, such as a least mean square (LMS) algorithm or a normalized LMS or recursive least square algorithm (RLS), for minimizing the time average of the power of the error signal e(k) from the subtractor  14  given by
 
 E [%           ( k )%□ 2 ]
 
where E[ ] denotes an expected value over a mean value of a value within the bracket [ ] and ∥e(k)∥ 2  is a square sum of e(k). In association with the updating of the tap coefficients of the FIR adaptive filter  41 , the coefficient {a k (i)} of the IIR filter  42  shown in the above equation (5) is also updated to {a k+1 (i)}.

     In the illustrative embodiment, shown in  FIG. 6 , optimum echo cancellation characteristics may be achieved even in the case of a small number of taps of the adaptive filter  15 , while the processing volume needs to be increased only to a lesser extent. 
     The filters of variable coefficients, such as the FIR adaptive filter  41  or the IIR filter  42 , configured for characteristics conversion, may be replaced by filters of fixed coefficients. 
       FIG. 7  shows an example of frequency characteristics of an FIR filter employed as a first characteristics conversion means, and  FIG. 8  shows a schematic structure of an echo removing apparatus employing a characteristics conversion filter of a fixed coefficient. That is, typical frequency characteristics of the fixed coefficient FIR filter  51  as the first characteristics conversion means of  FIG. 8  is shown in  FIG. 7 . 
     In  FIG. 8 , an FIR filter  51  operating as an inverse filter or a whitening filter with respect to the residual echo signal or the error signal E(k) is employed as a filter equivalent to the first characteristic conversion filter  21  of  FIG. 1 . 
     In  FIG. 8 , an N-tap FIR filter is employed as adaptive filter  15 . If the tap coefficients of the N-tap FIR filter are denoted by b k (i ) , where i=0, 1, . . . , N−1, its transfer function B k (k) is given by the equation: 
     
       
         
           
             
               
                 
                   
                     
                       B 
                       k 
                     
                     ⁡ 
                     
                       ( 
                       z 
                       ) 
                     
                   
                   = 
                   
                     
                       ∑ 
                       
                         i 
                         = 
                         0 
                       
                       
                         N 
                         - 
                         1 
                       
                     
                     ⁢ 
                     
                       
                         
                           b 
                           k 
                         
                         ⁡ 
                         
                           ( 
                           i 
                           ) 
                         
                       
                       ⁢ 
                       
                         z 
                         
                           - 
                           i 
                         
                       
                     
                   
                 
               
               
                 
                   ( 
                   10 
                   ) 
                 
               
             
           
         
       
     
     On the other hand, assuming that the transfer function W 1 (z) of, for example, an M-tap FIR filter  51 , equivalent to the first characteristic conversion filter  21  of  FIG. 4 , is represented by 
                       W   1     ⁡     (   z   )       =       1   -       A   k     ⁡     (   z   )         =     1   -       ∑     i   =   1     M     ⁢         a   k     ⁡     (   i   )       ⁢     z     -   i                       (   11   )               
In this case, a filter  52 , corresponding to the transfer function W 2 (z) of the second characteristic conversion filter  2  of  FIG. 4 , is an IIR filter whose transfer function W 2 (z) is given by the equation:
 
                       W   2     ⁡     (   z   )       =       1     1   -       A   k     ⁡     (   z   )           =     1     1   -       ∑     i   =   1     m     ⁢         a   k     ⁡     (   i   )       ⁢     z     -   i                         (   12   )               
Since the other structure of the filter is the same as that shown in  FIG. 4  or  6 , the corresponding portions are denoted by the same numerals and the corresponding description is omitted for simplicity.
 
     As the frequency characteristics of {1−A(z)} of the above equation ( 11 ), such characteristics which will suppress the low-range side energy while enhancing the high range side energy, as shown in  FIG. 7 , are employed. 
     In the illustrative embodiment of  FIG. 8 , the microphone input signal y(k) from the microphone  13  is filtered by the FIR adaptive filter  41  to derive a signal u(k) as shown by the equation (13): 
     
       
         
           
             
               
                 
                   
                     u 
                     ⁡ 
                     
                       ( 
                       k 
                       ) 
                     
                   
                   = 
                   
                     
                       y 
                       ⁡ 
                       
                         ( 
                         k 
                         ) 
                       
                     
                     - 
                     
                       
                         ∑ 
                         
                           i 
                           = 
                           1 
                         
                         M 
                       
                       ⁢ 
                       
                         
                           
                             a 
                             k 
                           
                           ⁡ 
                           
                             ( 
                             i 
                             ) 
                           
                         
                         ⁢ 
                         
                           y 
                           ⁡ 
                           
                             ( 
                             
                               k 
                               - 
                               i 
                             
                             ) 
                           
                         
                       
                     
                   
                 
               
               
                 
                   ( 
                   13 
                   ) 
                 
               
             
           
         
       
     
     On the other hand, the adaptive filter  15  receives the speaker output signal, which is the input signal at the terminal  11 , as a tap input signal, and generates an adaptive filter output signal given by: 
     
       
         
           
             
               
                 
                   
                     
                       u 
                       ^ 
                     
                     ⁡ 
                     
                       ( 
                       k 
                       ) 
                     
                   
                   = 
                   
                     
                       ∑ 
                       
                         i 
                         = 
                         0 
                       
                       
                         N 
                         - 
                         1 
                       
                     
                     ⁢ 
                     
                       
                         
                           b 
                           k 
                         
                         ⁡ 
                         
                           ( 
                           i 
                           ) 
                         
                       
                       ⁢ 
                       
                         x 
                         ⁡ 
                         
                           ( 
                           
                             k 
                             - 
                             1 
                           
                           ) 
                         
                       
                     
                   
                 
               
               
                 
                   ( 
                   14 
                   ) 
                 
               
             
           
         
       
     
     The subtractor  14  subtracts an adapter filter output signal of the equation (14) from the adaptive filter  15  from the above signal u(k) from the FIR adaptive filter  51  to derive an error signal e(k) represented by the equation (15):
 
 e ( k   0 = u ( k )− û ( k )  (15)
 
     This error signal e(k) is filtered by the above IIR filter  42  to produce an echo-reduced output signal z(k) represented by 
     
       
         
           
             
               
                 
                   
                     z 
                     ⁡ 
                     
                       ( 
                       k 
                       ) 
                     
                   
                   = 
                   
                     
                       e 
                       ⁡ 
                       
                         ( 
                         k 
                         ) 
                       
                     
                     + 
                     
                       
                         ∑ 
                         
                           i 
                           = 
                           1 
                         
                         M 
                       
                       ⁢ 
                       
                         
                           
                             a 
                             
                               k 
                               ⁢ 
                               
                                   
                               
                             
                           
                           ⁡ 
                           
                             ( 
                             i 
                             ) 
                           
                         
                         ⁢ 
                         
                           e 
                           ⁡ 
                           
                             ( 
                             
                               k 
                               - 
                               i 
                             
                             ) 
                           
                         
                       
                     
                   
                 
               
               
                 
                   ( 
                   16 
                   ) 
                 
               
             
           
         
       
     
     The tap coefficient {b k (i)} of the adaptive filter  15  is updated by a suitable adaptive algorithm for minimizing the time average of the power of the error signal e(k), as in the illustrative embodiment shown in  FIG. 6 . 
     With the embodiment of  FIG. 8 , the structure and the processing volume may be decreased, as compared to the embodiment shown in  FIG. 6 , in an amount corresponding to the adaptive processing of the characteristics conversion filter which may be omitted. 
     Next, in a system employing a sound generated encoding system, the coefficients of the characteristics conversion filter may be determined on the basis of sound generated encoding parameters. 
       FIG. 9  shows an illustrative embodiment of an echo removing apparatus employed on the decoder side of the sound generated encoding system. 
     In  FIG. 9 , filters  61 ,  62  are equivalent to the first characteristic conversion filter  21  and to the second characteristic conversion filter  22  of  FIG. 1 , respectively. 
     To an input terminal  63  are supplied parameters characteristic of sound generated and encoded sound generated signals encoded and transmitted by the encoder and received by the receiver. These encoded sound generated signals are decoded by sound generated decoder  64  into a speaker output signal x(k) as the generated sound signal or sound generated signal which is sent to the speaker  12  as sound generating means. 
     The parameters characteristic of sound generated, such as vocal tract parameters or a-parameters of VSELP, supplied from the input terminal  63 , are sent to a parameter converter  65  where they are converted into filter coefficients of the characteristics conversion filters  61 ,  62  for updating the filter coefficients of the filters  61 ,  62 . The coefficients of the filter  61  equivalent to the first characteristic conversion filter are converted into coefficients which will enable the whitening filter coefficients of whitening the input signals to be produced. On the other hand, the coefficients of the filter  62  are converted into coefficients which will enable opposite filter characteristics to be produced. 
     The microphone input signal y(k) from the microphone  13  is filtered by the filter  61  to derive the signal u(k) represented by the equation (17): 
     
       
         
           
             
               
                 
                   
                     u 
                     ⁡ 
                     
                       ( 
                       k 
                       ) 
                     
                   
                   = 
                   
                     
                       y 
                       ⁡ 
                       
                         ( 
                         k 
                         ) 
                       
                     
                     - 
                     
                       
                         ∑ 
                         
                           i 
                           = 
                           1 
                         
                         M 
                       
                       ⁢ 
                       
                         
                           
                             a 
                             k 
                           
                           ⁡ 
                           
                             ( 
                             i 
                             ) 
                           
                         
                         ⁢ 
                         
                           y 
                           ⁡ 
                           
                             ( 
                             
                               k 
                               - 
                               1 
                             
                             ) 
                           
                         
                       
                     
                   
                 
               
               
                 
                   ( 
                   17 
                   ) 
                 
               
             
           
         
       
     
     The adaptive filter  15 , receiving the speaker output signal x(k) from sound generated decoder  64  as the tap input signal, produces an adaptive filter output given by: 
     
       
         
           
             
               
                 
                   
                     
                       u 
                       ^ 
                     
                     ⁡ 
                     
                       ( 
                       k 
                       ) 
                     
                   
                   = 
                   
                     
                       ∑ 
                       
                         i 
                         = 
                         0 
                       
                       
                         N 
                         - 
                         1 
                       
                     
                     ⁢ 
                     
                       
                         
                           b 
                           k 
                         
                         ⁡ 
                         
                           ( 
                           i 
                           ) 
                         
                       
                       ⁢ 
                       
                         x 
                         ⁡ 
                         
                           ( 
                           
                             k 
                             - 
                             i 
                           
                           ) 
                         
                       
                     
                   
                 
               
               
                 
                   ( 
                   18 
                   ) 
                 
               
             
           
         
       
     
     The subtractor  14  subtracts the adaptive filter output from the signal u(k) from the signal u(k) from the FIR adaptive filter  61  to produce an error signal e(k) represented by the equation (19):
 
 e   9   k )= u ( k )− û ( k )  (19)
 
     The error signal e(k) from the subtractor  14  is filtered by the filter  62  to give an error-reduced signal z(k) represented by the equation: 
                     z   ⁡     (   k   )       =       e   ⁡     (   k   )       +       ∑     i   =   1     M     ⁢         a   k     ⁡     (   i   )       ⁢     e   ⁡     (     k   -   i     )                     (   20   )               
which is outputted at the terminal  16 .
 
     In the adaptive filter  15 , the tap coefficient {b k (i)} is updated by adaptive processing by any suitable adaptive algorithm for minimizing the time average of the power of the error signal e(k) from the subtractor  14 . On the other hand, the filter coefficients of the filters  61  and  62  are converted and updated into those of the whitening filter or the inverse filters thereof by a parameter converter  65 . 
     Next, an illustrative embodiment of employing the learning identification method or the normalized LMS (least mean square) method in the adaptive algorithm for the adaptive filter  15  for echo estimation in the above-described embodiments shown in  FIGS. 4 to 9 , is hereinafter explained. 
     In-the present illustrative embodiment, the smoothed input signal power value is employed as the tap input signal power of the equation employed for tap coefficients or filter coefficients of the learning identification method for producing echo removing characteristics or echo cancellation characteristics even in the case where the number of taps is smaller than the sound generated pitch period. 
     That is, in the illustrative embodiment of  FIG. 8 , if the usual learning identification method is used as the tap coefficient-adaptive algorithm of the N-tap FIR adaptive filter as the adaptive filter  15 , the equations for updating the N tap coefficients b k (i) into b k+1 (i) are: 
                         b     k   +   1       ⁡     (   i   )       =         b   k     +           δ   c     ⁢   ⁢     (   k   )     ⁢     x   ⁡     (     k   -   i     )           %   ⁢   ⁢     (   k   )     ⁢   %   ⁢     □   2         ⁢           ⁢   where   ⁢           ⁢   i       =   0       ,   1   ,       …   ⁢           ⁢   N     -   1             (   21   )               
and
 
                     %   ⁢   ⁢     (   k   )     ⁢   %   ⁢     □   2       =       ∑     i   =   0       N   -   1       ⁢     %   ⁢     □     x   ⁡     (     k   -   i     )         ⁢   %   ⁢     □   2                 (   22   )               
where μ is a constant known as a step gain parameter.
 
     However, if the tap length N of the FIR adaptive filter is shorter than the sound generated pitch period, as shown in  FIG. 10 , the denominator of the equation (21), that is the tap input signal power or the square sum, as calculated by the equation (22), is significantly fluctuated, as shown in  FIG. 11 . For example, the square sum, which is the power in the domain a corresponding to the tap length of  FIG. 10 , becomes larger, while the power in the domain b becomes smaller. If the input signal power or the square sum is fluctuated in this manner, the tap coefficient updated by the equation (21) the tap coefficients updated by the equation (21) are fluctuated, thus occasionally making it impossible to produce stable echo removing or suppression characteristics. Thus the denominator of the equation (21), that is the input signal power or the square sum calculated by the equation (22), is replaced by a power value smoothed by a suitable method, that is a smoothed value of the input signal power P x (k), for realizing stable echo removing or suppression characteristics. 
     If such smoothed input signal power value P x (k) is employed, tap coefficient updating is performed in accordance with the following equation (23): 
     
       
         
           
             
               
                 
                   
                     
                       
                         b 
                         
                           k 
                           + 
                           1 
                         
                       
                       ⁡ 
                       
                         ( 
                         i 
                         ) 
                       
                     
                     = 
                     
                       
                         b 
                         k 
                       
                       + 
                       
                         
                           
                             δ 
                             c 
                           
                           ⁢ 
                           ⁢ 
                           
                             ( 
                             k 
                             ) 
                           
                           ⁢ 
                           
                             x 
                             ⁡ 
                             
                               ( 
                               
                                 k 
                                 - 
                                 1 
                               
                               ) 
                             
                           
                         
                         
                           
                             P 
                             x 
                           
                           ⁡ 
                           
                             ( 
                             k 
                             ) 
                           
                         
                       
                     
                   
                   , 
                   
                     i 
                     = 
                     0 
                   
                   , 
                   1 
                   , 
                   … 
                   ⁢ 
                   
                       
                   
                   , 
                   
                     N 
                     - 
                     1 
                   
                 
               
               
                 
                   ( 
                   23 
                   ) 
                 
               
             
           
         
       
     
     An illustrative example of the method for calculating the smoothed value of the input signal power P x (k) in the equation (23) is to find a square sum value of the number of samples L sufficiently larger than the number of taps N in order to find a value normalized over N samples, that is to execute the calculation denoted by 
                       P   k     ⁡     (   k   )       =       N   L     ⁢       ∑     i   =   0       L   -   1       ⁢     %   ⁢     □     x   ⁡     (     k   -   i     )         ⁢   %   ⁢     □   2                   (   24   )               
Another method for calculating the smoothed input signal power P x (k) is to effect smoothing by a time constant longer than the pitch period of the input signal, that is to execute the calculation denoted by
   P   x ( k )=&amp;□ P   x ( k− 1)+(1−&amp;           %□ x ( k )%□ 2   
(25)
 
In the equation (25), λ is a constant such that 0&lt;λ&lt;1, with the corresponding time constant being 1/(1−λ).

     By employing the smoothed power, that is the smoothed input signal power P x (k), variations in the filter coefficients or tap coefficients may be suppressed for achieving stable echo removing or suppression characteristics. 
     Referring to  FIG. 12 , the echo cancellation characteristics in the case of employing the above-described structure of the illustrative embodiment will be hereinafter explained. 
     In the graph of  FIG. 12 , the number of taps of the FIR filter, as the adaptive filter  15 , is taken on the abscissa and the amount of echo cancellation ERLE is taken on the ordinate. The amount of echo cancellation is defined by the following equation (26): 
     
       
         
           
             
               amount 
               ⁢ 
               
                   
               
               ⁢ 
               of 
               ⁢ 
               
                   
               
               ⁢ 
               echo 
               ⁢ 
               
                   
               
               ⁢ 
               cancellation 
               ⁢ 
               
                   
               
               ⁢ 
               
                 ( 
                 ERLE 
                 ) 
               
             
             ⁢ 
             
                 
             
             = 
             
               
                 
                   time 
                   ⁢ 
                   
                       
                   
                   ⁢ 
                   average 
                   ⁢ 
                   
                       
                   
                   ⁢ 
                   of 
                   ⁢ 
                   
                       
                   
                   ⁢ 
                   echo 
                   ⁢ 
                   
                       
                   
                   ⁢ 
                   cancelor 
                   ⁢ 
                   
                       
                   
                   ⁢ 
                   output 
                   ⁢ 
                   
                       
                   
                   ⁢ 
                   voltage 
                 
                 ⁢ 
                 
                     
                 
               
               
                 time 
                 ⁢ 
                 
                     
                 
                 ⁢ 
                 average 
                 ⁢ 
                 
                     
                 
                 ⁢ 
                 of 
                 ⁢ 
                 
                     
                 
                 ⁢ 
                 microphone 
                 ⁢ 
                 
                     
                 
                 ⁢ 
                 input 
                 ⁢ 
                 
                     
                 
                 ⁢ 
                 voltage 
               
             
           
         
       
     
     The echo canceler output voltage of the equation ( 26 ) is the power of the signal z(k) taken out at the terminal  16 , while the microphone input voltage is the power of the microphone input signal y(k) from the microphone  13 . 
     In  FIG. 12 , a curve a stands for the amount of echo cancellation in the case of using a characteristic conversion filter FIR filter  51 , such as a fixed coefficient filter with the number CL of taps M equal to  12 , and a filter  52  functioning as its inverse filter, while a curve  b  stands for the amount of echo cancellation in the case of using a conventional structure not employing the filters  51  and  52 . 
     It is seen from  FIG. 12  that, for a domain corresponding to a smaller number of taps of the adaptive filter  15 , the echo cancellation characteristics can be significantly improved by adding the characteristics conversion filters  51  and  52 . As such a smaller processing volume suffices. That is, even if the characteristics conversion filters are added, a smaller processing volume suffices if the number of taps of the adaptive filter  15  is less than a tenth of that in the case of a conventional echo removing apparatus not provided with characteristics conversion filters. 
     Thus, with a smaller number of taps of the FIR adaptive filters, echo cancellation characteristics with higher effects may be achieved, while the processing volume required for achieving echo cancellation characteristics comparable to those of the echo removing apparatus may be reduced. 
     The present invention is not limited to the above-described embodiments. For example, although the basic structure of  FIG. 1  is implemented in the illustrative embodiments of  FIGS. 6 to 9 , the basic structure of  FIG. 5  may be implemented in a similar manner. In addition, the present invention may be applied to variety of sound generated communication terminals, in addition to the portable telephone. Sound generating means or sound generated collecting means are not limited to the speaker or to the microphone. In addition, the filter coefficient of the FIR adaptive filter may be estimated not only by the learning identification method but by a variety of other adaptive algorithms.