Adaptive digital filter with high speed and high precision coefficient sequence generation

An adaptive digital filter includes a finite impulse response digital filter for performing a convolution calculation by multiplying a code sequence with a coefficient sequence, and a coefficient calculation unit for performing the updating calculation of the coefficient sequence only when the calculation result of the finite impulse response digital filter belongs to a predetermined numerical value region other than numerical value regions which are defined in advance in correspondence with code values of the code sequence. Thus, a coefficient sequence is generated at high speed and with high precision without making the calculation result of the coefficient sequence diverge even for a code sequence in which considerable intersymbol interference has occurred due to disturbance on the transmission path, and the intersymbol interference can be removed.

BACKGROUND OF THE INVENTION 
(1) Field of the Invention 
The present invention relates to an adaptive digital filter and, more 
particularly, to an adaptive digital filter which is suitable for an 
adaptive equalizer or the like in a digital communication apparatus that 
uses vestigial sideband (to be abbreviated as "VSB" hereinafter) 
modulation, quadrature amplitude modulation (to be abbreviated as "QAM" 
hereinafter), or the like. 
(2) Description of the Background Art 
Nowadays, attempts are being made worldwide to realize television 
broadcasting using a digital transmission technique. When such television 
broadcasting is realized as terrestrial broadcasting, intersymbol 
interference due to reflected waves, i.e., ghost, poses a serious problem, 
and an adaptive digital filter is used for eliminating this ghost. 
The adaptive digital filter means a digital filter that performs 
convolution calculation processing by appropriately controlling a 
coefficient sequence to be multiplied with a code sequence in accordance 
with the signal state of the code sequence input as an input signal, and 
realizes appropriate filter characteristics corresponding to the signal 
state of an input signal that changes with time. 
The adaptive digital filter is normally constituted using a finite impulse 
response feedforward digital filter (to be referred to as an "FIR digital 
filter" hereinafter) and an infinite impulse response feedback digital 
filter (to be referred to as an "IIR digital filter hereinafter). 
In the following description, a variable n indicates a natural number which 
time-sequentially corresponds to codes of a code sequence to be input to 
the adaptive digital filter. 
FIG. 1 shows the arrangement of a conventional typical adaptive digital 
filter. The conventional adaptive digital filter shown in FIG.1 comprises 
an FIR digital filter 1 that performs a convolution calculation by 
multiplying an input code sequence U(n) u(n+K1), u(n+K1-1), . . . , u(n)! 
with a coefficient sequence C.sub.k (n) c.sub.-K1 (n), c.sub.-K1+1 (n), . 
. . , c.sub.0 (n)!, an IIR digital filter 2 that performs a convolution 
calculation by multiplying an estimated value sequence W(n) w(n), . . . , 
w(n-K2-1), w(n-K2)! input from a slicer 4 (to be described later) with a 
coefficient sequence C.sub.k (n) c.sub.1 (n), . . . , c.sub.K 2 1 (n), 
c.sub.K2 (n)!, an adder 3 for adding the calculation results from the FIR 
and IIR digital filters 1 and 2, a slicer 4 for outputting an estimated 
value w(n) on the basis of the sum output from the adder 3, a subtracter 5 
for subtracting the sum output from the adder 3 from the estimated value 
w(n) output from the slicer 4 and outputting an error value e(n), and a 
multiplier 7 for multiplying the error value e(n) calculated by the 
subtracter 5 with a convergence coefficient .alpha. to generate the 
coefficient sequence C.sub.k (n) c.sub.K1 (n), c.sub.K1+1 (n), . . . , 
c.sub.0, c.sub.1 (n), . . . , c.sub.K2 1 (n), c.sub.K2 (n)!. Note that the 
output from the adder 3 serves as the output from this conventional 
adaptive digital filter, and this calculation result is output as an 
equalization signal y(n) to an external device. 
The FIR digital filter 1 shown in FIG.1 comprises unit delay elements 40 
cascade-connected between adjacent taps, multipliers 41 for respectively 
multiplying the code values of the code sequence U(n) u(n+K1), u(n+K1-1), 
. . . , u(n)! appearing at the respective taps with the coefficient values 
of the coefficient sequence C.sub.k (n) c.sub.K1 (n), c.sub.K1+1 (n), . . 
. , c.sub.0 (n)!, integrators 42 for respectively integrating the products 
output from the multipliers 41 of the respective taps, multipliers 43 for 
respectively multiplying code values u(x) {x=n+K1, n+K1-1, . . . , n} 
appearing at the respective taps with the integration results output from 
the integrators 42, and an adder 44 for receiving the products output from 
the multipliers 43 of the respective taps, and adding the products. Note 
that a constant K1 indicates a value obtained by subtracting 1 from the 
number of taps of the FIR digital filter 1. 
The IIR digital filter 2 also has substantially the same arrangement as 
that of the above-mentioned FIR digital filter 1, except that the input 
code value is the estimated value w(n) output from the slicer 4, and the 
coefficient values of the coefficient sequence C.sub.k (n) to be 
multiplied with the code values of the respective taps are c.sub.1 (n), . 
. . , c.sub.K2 1 (n), c.sub.K2 (n). Note that a constant K2 indicates the 
number of taps of the IIR digital filter 2. 
The operation of the conventional adaptive digital filter with the above 
arrangement will be described in detail below with reference to FIG. 1. 
Referring to FIG. 1, when a code sequence U(n) u(n+K1), u(n+K1-1), . . . , 
u(n)! is input to the conventional adaptive digital filter, the FIR 
digital filter 1 performs convolution calculation processing by 
multiplying code values u(x) {x=n+K1, n+K1-1, . . . , n} of the code 
sequence U(n) with coefficient values c.sub.k (n) {k=-K1, -K1+1, . . . , 
0). 
More specifically, the code values of the input code sequence U(n) are 
delayed by the unit delay elements 40 constituting the FIR digital filter 
1, code values u(n+K1), u(n+K1-1), . . . , u(n+1) appear in turn from the 
taps on the front stage side, and a code value u(n) appears at the tap of 
the final stage. The multipliers 41 multiply code values u(x) appearing at 
the respective taps with the coefficient values c.sub.k (n). 
The products output from the multipliers 41 at the respective taps are 
integrated by the integrators 42, and the multipliers 43 multiply the 
integration results output from the integrators 42 with the code values 
appearing at the respective taps. The adder 44 adds the products output 
from the multipliers 43 of the respective taps, and outputs the sum as a 
calculation result y.sub.1 (n) of the FIR digital filter 1. 
On the other hand, the IIR digital filter 2 receives an estimated value 
sequence W(n) w(n), . . . , w(n-K2-1), w(n-K2)! (to be described later), 
and similarly performs convolution calculation processing by respectively 
multiplying estimated values w(n), w(n-K2-1), w(n-K2) appearing at the 
respective taps with coefficient values c.sub.1 (n), . . . , c.sub.K2 1 
(n), c.sub.K2 (n). The adder 44 constituting the final stage of the IIR 
digital filter 2 adds the calculation results of the respective taps, and 
outputs the sum as a calculation result y.sub.2 (n) of the IIR digital 
filter 2. 
The adder 3 receives the calculation results y.sub.1 (n) and y.sub.2 (n) 
from the FIR and IIR digital filters 1 and 2, and adds them to output a 
sum y(n). 
The slicer 4 receives the sum y(n) output from the adder 3, and outputs an 
estimated value w(n) which is most approximate to the sum y(n). 
The estimated value w(n) output from the slicer 4 represents a value 
obtained when the receiver estimates a code value of a code sequence 
transmitted from the transmitter. More specifically, the code sequence 
U(n) to be received by the receiver is modulated as a result of various 
types of disturbance on the transmission path, and becomes different from 
that transmitted from the transmitter. The estimated value w(n) represents 
a code value which is estimated to be received by the receiver if there is 
no disturbance on the transmission path, and the slicer 4 selects and 
outputs, as the estimated value w(n), a code value closest to an actually 
received code value from normal code values, which are agreed upon in 
advance between the transmitter and receiver. 
The subtracter 5 then subtracts the estimated value w(n) output from the 
slicer 4 from the sum y(n) output from the adder 3, and outputs an error 
value e(n). More specifically, the subtracter 5 calculates an error 
generated in the code value y(n) to be output from the receiver due to 
disturbance on the transmission path that acts on the code value 
transmitted from the transmitter. 
The multiplier 7 generates a coefficient sequence C.sub.k (n) by 
multiplying the error value e(n) output from the subtracter 5 with a 
convergence coefficient .alpha., and supplies coefficient values of the 
generated coefficient sequence C.sub.k (n) to the multipliers 41 
constituting the FIR and IIR digital filters 1 and 2. At this time, 
coefficient values C.sub.-Kl (n); C.sub.-Kl+l (n);. . . ; C.sub.0 ; 
C.sub.l (n);. . . C.sub.K2-1 (n); C.sub.K2 (n) of the coefficient sequence 
C.sub.k (n) are calculated and generated so that the error value e(n) 
output from the subtracter 5 converges to a minimum value. 
The above-mentioned operation can be expressed by formula (1) below: 
##EQU1## 
The method of generating the coefficient sequence C.sub.k (n) when the 
adaptive digital filter is used as an adaptive equalizer in a data 
transmission system will be explained below. 
Assume that a code value u(n) is an input signal received from the 
transmission path, and the input signal u(n) is subjected to disturbance 
by a disturbance signal on the transmission path. The disturbance amount 
of the disturbance signal on the input signal u(n) depends on the 
characteristics of the transmission path. More specifically, the 
disturbance signal is modulated by the characteristic function of the 
transmission path, and influences the signal level of a code sequence on 
the transmission path. 
Therefore, in order to remove the disturbance signal components 
superimposed on the input signal u(n) on the transmission path, the 
receiver need only realize the inverse function of the characteristic 
function of the transmission path, and process the input signal u(n). For 
this purpose, the coefficient sequence C.sub.k (n) is generated so that 
the characteristics of the adaptive digital filter realize the inverse 
function of the characteristic function of the transmission path with 
respect to the disturbance signal. As a result, the coefficient sequence 
C.sub.k (n) is sequentially updated to minimize the error value e(n), and 
finally converges to a value from which the disturbance on the 
transmission path is removed. 
As a typical algorithm for generating the coefficient sequence C.sub.k (n) 
to realize the inverse function of the characteristic function of the 
transmission path, an LMSE (Least Mean Square Error Method) algorithm is 
known. This algorithm pays attention to a square mean error D, and 
generates the coefficient sequence C.sub.k (n) to minimize the square mean 
error D. The square mean error D is a quantity defined by the following 
formula (2): 
EQU D={e(n)}.sup.2 ( 2) 
Using the square mean error D defined as described above, each coefficient 
c.sub.k of the coefficient sequence C.sub.k (n) is calculated from 
formulas (3a) and (3b) below. More specifically, the coefficient value 
c.sub.k is repetitively calculated until the error value e(n) in formulas 
(3a) and (3b) converges to a minimum value, and all the coefficient values 
of the coefficient sequence C.sub.k (n) are finally obtained. 
##EQU2## 
Formula (3a) is one for calculating the coefficient sequence to be applied 
to the FIR digital filter 1, and formula (3b) is one for calculating the 
coefficient sequence to be applied to the IIR digital filter 2. Note that 
a constant .alpha. indicates a convergence coefficient, and is set to be a 
small, positive value. 
On the other hand, the error value e(n) is given by formula (4) below since 
it is the difference between the estimated value w(n) and the output value 
y(n): 
EQU I(I)=I(I) -I(I) (4) 
Formula (4) is calculated in the process of calculating the coefficient 
sequence C.sub.k (n) to minimize the error value e(n), and the following 
two different coefficient sequence calculation methods are known which 
differ in the way of defining the estimated value w(n) in formula (4). 
A) Training Sequence Method 
This calculation method can be used when a portion of a code sequence to be 
transmitted includes a reference signal of a pattern corresponding to a 
predetermined code value, and the estimated value w(n) is calculated based 
on this training sequence to update a coefficient sequence. 
B) Symbol Decision Method 
In this calculation method, the actually received code value is defined as 
the one with maximum likelihood, and is used as an estimated value w(n). 
As one of characteristics required for a television receiver, a reception 
image is to be obtained within a short period of time after the power 
switch is turned on or a channel is switched. This period of time depends 
on the convergence time of the calculation of the coefficient sequence 
C.sub.k (n). Therefore, in order to improve the characteristics of the 
television receiver, the calculation time of the coefficient sequence 
C.sub.k (n) must be shortened. 
From this point of view, merits and demerits of the training sequence 
method and the symbol decision method of the LMSE algorithm will be 
explained. 
According to the training sequence method, since a correct estimated value 
w(n) is guaranteed, theoretically, the calculation result always converges 
without divergence. However, if the ratio of the training sequence in a 
code sequence to be transmitted increases, the code transfer efficiency 
lowers. For this reason, in practice, the training sequence can only be 
included in a code sequence at a very small ratio (e.g., about 1/500) with 
respect to all the codes to be transmitted. For this reason, much time is 
required to calculate the coefficient sequence. 
On the other hand, according to the symbol decision method, since the 
calculation can be made using all the codes to be transmitted, the 
calculation result converges within a very short period of time. However, 
if the estimated value w(n) has a large difference from the actually 
transmitted code due to disturbance such as intersymbol interference, the 
calculation result of the coefficient sequence C.sub.k (n) readily 
diverges. 
For example, when the received code signal is the one that is expressed by 
a constellation free from any intersymbol interference, as shown in FIG. 
2B, since no error occurs upon discrimination of the code value, the 
calculation result of the coefficient sequence converges at high speed 
using the symbol decision method. 
However, when the received code signal is the one expressed by a 
constellation suffering intersymbol interference, as shown in FIG. 2C, 
since errors frequently occur upon discrimination of the code value, the 
calculation result of the coefficient sequence C.sub.k (n) does not 
converge, and may diverge in the worst case. 
Note that the constellation shown in FIG. 2A is that of a signal free from 
any intersymbol interference, and is an example of an ideal constellation. 
As described above, according to the adaptive digital filter using the LMSE 
algorithm based on the training sequence method, much time is required 
until the calculation result of the coefficient sequence C.sub.k (n) 
converges. On the other hand, according to the adaptive digital filter 
using the LMSE algorithm based on the symbol decision method, considerable 
intersymbol interference cannot be coped with. 
SUMMARY OF THE INVENTION 
The present invention has been made in consideration of the above-mentioned 
problems and has as its object to provide an adaptive digital filter that 
can remove intersymbol interference by generating a coefficient sequence 
at high speed and with high precision, even for a code sequence suffering 
considerable intersymbol interference, without divergence of the 
calculation result of the coefficient sequence. 
In order to achieve the above object, the present invention comprises the 
following arrangement. 
According to the first aspect of the present invention, there is provided 
an adaptive digital filter for removing intersymbol interference generated 
in a code sequence in a transmission path, comprising finite impulse 
response digital filter means for performing a convolution calculation by 
multiplying the code sequence with a coef ficient sequence, and 
coefficient calculation means for, when a calculation result from the 
finite impulse response digital filter means belongs to a predetermined 
numerical value region other than a plurality of numerical value regions 
which are defined in advance in correspondence with code values of the 
code sequence, generating the coefficient sequence on the basis of a 
predetermined value which belongs to one of the plurality of numerical 
value regions closest to the predetermined numerical value region. 
According to the second aspect of the present invention, there is provided 
an adaptive digital filter for removing intersymbol interference generated 
in a code sequence in a transmission path, comprising finite impulse 
response digital filter means for performing a convolution calculation by 
multiplying the code sequence with a coefficient sequence, addition means 
for receiving a calculation result from the finite impulse response 
digital filter means as one input value, and adding another input value to 
the one input value, coefficient calculation means for, when a sum output 
from the addition means belongs to a predetermined numerical value region 
other than a plurality of numerical value regions which are defined in 
advance in correspondence with code values of the code sequence, 
generating the coefficient sequence on the basis of a predetermined value 
which belongs to one of the plurality of numerical value regions closest 
to the predetermined numerical value region, and for, when the sum output 
from the addition means belongs to one of the plurality of numerical value 
regions, generating the coefficient sequence on the basis of a 
predetermined value that belongs to the numerical value region to which 
the sum belongs, and infinite impulse response digital filter means for 
performing a convolution calculation by multiplying a sequence of the 
predetermined values with the coefficient sequence, and supplying the 
calculation result to the addition means as the other input value. 
In an adaptive digital filter according to the third aspect of the present 
invention, the coefficient calculation means of the adaptive digital 
filter of the first aspect comprises a discrimination unit for 
discriminating the numerical value region to which the calculation result 
of the finite impulse response digital filter means belongs from the 
plurality of numerical value regions which are defined in advance in 
correspondence with the code values of the code sequence, outputting a 
predetermined value that belongs to the discriminated numerical value 
region, and outputting an identification signal for identifying whether or 
not the calculation result belongs to the predetermined numerical value 
region, an error calculation unit for calculating a difference between the 
calculation result and the predetermined value, and outputting an error 
value, a selection unit for selecting and outputting the error value on 
the basis of the identification signal, and a coefficient sequence 
generation unit for generating the coefficient sequence on the basis of an 
output from the selection unit. 
In an adaptive digital filter according to the fourth aspect of the present 
invention, the coefficient calculation means of the adaptive digital 
filter of the second aspect comprises a discrimination unit for 
discriminating the numerical value region to which the sum output from the 
addition means belongs from the plurality of numerical value regions which 
are defined in advance in correspondence with the code values of the code 
sequence, outputting a predetermined value that belongs to the 
discriminated numerical value region, and outputting an identification 
signal for identifying whether or not the sum belongs to the predetermined 
numerical value region, an error calculation unit for calculating a 
difference between the sum and the predetermined value, and outputting an 
error value, a selection unit for selecting and outputting the error value 
on the basis of the identification signal, and a coefficient sequence 
generation unit for generating the coefficient sequence on the basis of an 
output from the selection unit. 
In an adaptive digital filter according to the fifth aspect of the present 
invention, the coefficient calculation means of the adaptive digital 
filter of the first or second aspect updates a central value in the 
coefficient sequence on the basis of an error value of the calculation 
result from the finite impulse response digital filter means with respect 
to a training sequence included in the code sequence during an input 
period of the training sequence. 
In an adaptive digital filter according to the sixth aspect of the present 
invention, the coefficient sequence generation unit of the adaptive 
digital filter of the third or fourth aspect updates a central value in 
the coefficient sequence on the basis of an error value of the calculation 
result from the finite impulse response digital filter means with respect 
to a training sequence included in the code sequence during an input 
period of the training sequence. 
In an adaptive digital filter according to the seventh aspect of the 
present invention, the selection unit of the adaptive digital filter of 
the third or fourth aspect selects the error value on the basis of 
convergence/non-convergence of a training sequence during an input period 
of the training sequence included in the code sequence, independently of 
the identification signal. 
In an adaptive digital filter according to the eighth aspect of the present 
invention, the selection unit of the adaptive digital filter of the third 
or fourth aspect selects the error value on the basis of 
convergence/non-convergence of a training sequence during an input period 
of the training sequence included in the code sequence, independently of 
the identification signal, and the coefficient sequence generation unit 
updates a central value in the coefficient sequence on the basis of an 
error value of the calculation result from the finite impulse response 
digital filter means with respect to a training sequence during an input 
period of the training sequence included in the code sequence. 
In an adaptive digital filter according to the ninth aspect of the present 
invention, the predetermined numerical value region of the adaptive 
digital filter of the first or second aspect is a numerical value region 
having a value larger than a value that belongs to the numerical value 
region defined in correspondence with a code value with a maximum absolute 
value. 
In an adaptive digital filter according to the 10th aspect of the present 
invention, the predetermined numerical value region of the adaptive 
digital filter of the third or fourth aspect is a numerical value region 
having a value larger than a value that belongs to the numerical value 
region defined in correspondence with a code value with a maximum absolute 
value. 
In an adaptive digital filter according to the 11th aspect of the present 
invention, the coefficient sequence generation unit of the adaptive 
digital filter of the third or fourth aspect generates the coefficient 
sequence to minimize an error value. 
The functions of the adaptive digital filters according to the first to 
11th aspects of the present invention will be described below. 
With the adaptive digital filter according to the first aspect of the 
present invention, when the calculation result of the finite impulse 
response digital filter means with respect to an input code sequence does 
not belong to any of the plurality of numerical value regions which are 
defined in advance in correspondence with the code values of the code 
sequence, and belongs to the predetermined numerical value region, a 
coefficient sequence is generated on the basis of a predetermined value 
that belongs to a numerical value region closest to the predetermined 
numerical value region. At this time, when the numerical value level of 
the code value increases due to disturbance, most of the code values 
belonging to the predetermined regions are those belonging to the 
numerical value region closest to the predetermined numerical value 
region. Hence, it is highly probable that the predetermined value is an 
original value of the code value whose numerical value level has increased 
due to the disturbance. Therefore, by generating the coefficient sequence 
based on the predetermined value, a high-precision coefficient sequence 
can be obtained. 
With the adaptive digital filter according to the second aspect of the 
present invention, when the sum of the calculation result of the finite 
impulse response digital filter means and the calculation result of the 
infinite impulse response digital filter means with respect to an input 
code sequence does not belong to any of the plurality of numerical value 
regions which are defined in advance in correspondence with the code 
values of the code sequence, and belongs to the predetermined numerical 
value region, a coefficient sequence is generated on the basis of a 
predetermined value that belongs to a numerical value region closest to 
the predetermined numerical value region. At this time, when the numerical 
value level of the code value increases due to disturbance, most of the 
code values belonging to the predetermined regions are those belonging to 
the numerical value region closest to the predetermined numerical value 
region. Hence, it is highly probable that the predetermined value is an 
original value of the code value whose numerical value level has increased 
due to disturbance. Therefore, by generating the coefficient sequence 
based on the predetermined value, a high-precision coefficient sequence 
can be obtained. Also, since the calculation result obtained by the 
convolution calculation of the infinite impulse response digital filter 
means by multiplying a predetermined value sequence in the numerical value 
region with a coefficient sequence is fed back to and added to the 
calculation result of the finite impulse response digital filter means, 
amplification of linear noise can be suppressed in a process of generating 
a coefficient sequence by the coefficient calculation means. 
With the adaptive digital filter according to the third aspect of the 
present invention, when the calculation result of the finite impulse 
response digital filter means belongs to the predetermined numerical value 
region, the discrimination unit outputs a predetermined value that belongs 
to a numerical value region of the plurality of numerical value regions 
closest to the predetermined numerical value region, and outputs an 
identification signal for identifying whether or not the calculation 
result belongs to the predetermined numerical value region. The error 
calculation unit outputs an error value of the calculation result from the 
predetermined value. When the calculation result belongs to the 
predetermined numerical value region, the selection unit is controlled by 
the identification signal to select the error value, and supplies the 
error value to the coefficient sequence generation unit. The coefficient 
sequence generation unit generates a coefficient sequence on the basis of 
the error value of the calculation result from the predetermined value in 
the predetermined numerical value region. 
With the adaptive digital filter according to the fourth aspect of the 
present invention, when the sum output from the addition means belongs to 
the predetermined numerical value region, the discrimination unit outputs 
a predetermined value that belongs to a numerical value region of the 
plurality of numerical value regions closest to the predetermined 
numerical value region, and outputs an identification signal for 
identifying whether or not the calculation result belongs to the 
predetermined numerical value region. The error calculation unit outputs 
an error value of the calculation result from the predetermined value. 
When the calculation result belongs to the predetermined numerical value 
region, the selection unit is controlled by the identification signal to 
select the error value, and supplies the error value to the coefficient 
sequence generation unit. The coefficient sequence generation unit 
generates a coefficient sequence on the basis of the error value of the 
calculation result from the predetermined value in the predetermined 
numerical value region. 
With the adaptive digital filter according to the fifth and sixth aspects 
of the present invention, since only the central value of a coefficient 
sequence is updated based on the training sequence included in the 
coefficient sequence, the calculation convergence time of the coefficient 
sequence can be shortened. 
With the adaptive digital filter according to the seventh aspect of the 
present invention, since the convergence/non-convergence of the training 
sequence included in a code sequence is monitored, a coefficient sequence 
can be generated using code values which suffer less disturbance, and the 
calculation convergence time of the coefficient sequence can be further 
shortened. 
With the adaptive digital filter according to the eighth aspect of the 
present invention, since the convergence/non-convergence of the training 
sequence included in a code sequence is monitored to generate a 
coefficient sequence using code values which suffer less disturbance, and 
only the central value of the coefficient sequence is updated based on the 
training sequence, the calculation convergence time of the coefficient 
sequence can be further shortened. 
With the adaptive digital filter according to the ninth and 10th aspects of 
the present invention, since the predetermined numerical value region is a 
numerical value region having a value larger than that belonging to a 
numerical value region which is defined in correspondence with a code 
value with a maximum absolute value, when a code value which is to 
originally belong to the numerical value region which is defined in 
correspondence with the code value with the maximum absolute value belongs 
to the predetermined numerical value region under the influence of 
disturbance, the numerical value region to which the disturbed code value 
is to originally belong can be easily estimated with high precision. 
With the adaptive digital filter according to the 11th aspect of the 
present invention, since a coefficient sequence is generated to minimize 
the error value, the coefficient sequence output that can remove 
disturbance can be obtained.

DESCRIPTION OF THE PREFERRED EMBODIMENTS 
An adaptive digital filter according to the preferred embodiments of the 
present invention will be described hereinafter with reference to the 
accompanying drawings. 
(First Embodiment) 
FIG. 3 is a block diagram showing the arrangement of an adaptive digital 
filter according to the first embodiment of the present invention. FIG. 3 
shows an example of the arrangement to be applied to transmission path 
waveform distortion correction. The same reference numerals in FIG. 3 
denote the same parts as in the conventional digital filter shown in FIG. 
1, and a detailed description thereof will be omitted. The adaptive 
digital filter of this embodiment has substantially the same arrangement 
as that of the conventional digital filter shown in FIG. 1, except that a 
slicer 4A is arranged in place of the slicer 4 shown in FIG. 1, a 
reception signal/error selection unit 6 is arranged at the output side of 
a subtracter 5, and an integrator 42A replaces the integrator at the 
center tap of the FIR digital filter 1. 
Note that the integral operation of the integrator 42A is controlled by a 
control signal S, so that the integrator 42A is activated to perform an 
integral operation when the signal state of the control signal S is a 
logic value "1"; the integrator 42A is deactivated to hold the previous 
integrated value when the signal state of the control signal S is a logic 
value "0". In the description of this embodiment, assume that the signal 
state of the control signal S is fixed to the logic value "0", and the 
center tap coefficient is updated together with the coefficients of other 
taps. 
The slicer 4A receives a sum y(n) output from an adder 3, and calculates 
and outputs an estimated value w(n) which is most approximate to the sum 
y(n). Upon calculating the estimated value w(n), the slicer 4A 
discriminates to which one of a numerical value region that is defined to 
have a normal code value free from any modulation due to disturbance as 
the central value and a predetermined numerical value region (to be 
described later) the sum y(n) output from the adder 3 belongs. 
The slicer 4A outputs, as the estimated value w(n), a code value as the 
central value of the numerical value region with respect to a group of 
sums y(n) that belong to the numerical value region defined to have the 
normal code value as the central value. Also, the slicer 4A discriminates 
whether or not the sum y(n) output from the adder 3 belongs to the 
predetermined numerical value region (to be described later), and outputs 
an identification signal that reflects the discrimination result. 
In the following description, a "numerical value region" means the 
numerical value region defined to have the normal code value as the 
central value, and is clearly distinguished from the "predetermined 
numeral value region" to be described later. The "numerical value region" 
and the "predetermined numeral value region" respectively correspond to 
numerical value regions G.sub.1 to G.sub.8 and regions X shown in FIG. 4. 
The discrimination of the numerical value regions by the slicer 4A of the 
digital filter of this embodiment will be explained in more detail below 
with reference to FIG. 4. The above-mentioned estimated value w(n) is one 
of central values W.sub.i (i=1, 2, . . . , 8) of the numerical value 
regions G.sub.1 to G.sub.8, and is a value to be originally assumed by a 
code value u(n) of the received code sequence. Therefore, when the input 
code sequence is free from any disturbance on the transmission path, the 
received code value u(n) matches one of the estimated value w(n), i.e., 
one of the central values W.sub.i (i=1, 2, . . . , 8) of the numerical 
value regions G.sub.1 to G.sub.8. 
However, when the input code sequence suffers disturbance on the 
transmission path, the signal level of the code sequence varies, and these 
values do not match with each other. Therefore, in this case, the original 
code value, i.e., the transmission code value must be estimated from the 
received code value u(n) suffering disturbance. 
As described above, the numerical value regions are defined in advance to 
have normal code values as their central values. If the received code 
value u(n) belongs to one of these numerical value regions, this code 
value u(n) is determined as the one expressed by the central value of the 
numerical value region to which the code value belongs, and the central 
value of this numerical value region is determined as the estimated value 
w(n) of the received code value u(n). 
The number of numerical value regions is determined by the number of levels 
that can be assumed by the code value in question. In the case of an 8 VSB 
code signal shown in FIG. 4 described above, the number of levels of the 
code value is 8, and central values W.sub.1 to W.sub.8 are respectively 
present at the centers of the numerical value regions G.sub.1 to G.sub.8. 
Note that the central values may be determined as average levels of the 
numerical value regions G.sub.1 to G.sub.8, but may be determined at 
appropriate positions of the respective numerical value regions in 
correspondence with the characteristics of an input signal. Also, the 
numerical value range of each of the numerical value regions G.sub.1 to 
G.sub.8 may be appropriately determined in correspondence with the 
characteristics of the input signal. 
Numerical value regions indicated by regions X in FIG. 4 are those having 
values larger than the maximum numerical value region G.sub.8 or smaller 
than the minimum numerical value region G.sub.1. More specifically, when a 
signal with an excessively large amplitude (absolute value) is input due 
to, e.g., disturbance, the code value of this signal belongs to one of 
these regions X. 
The constellation shown in FIGS. 2A to 2C is that of an 8VSB code signal, 
and when an intersymbol interference has occurred due to, e.g., 
disturbance, the signal level of the code value varies and deviates from 
the central value of the corresponding numerical value region, as shown in 
FIG. 2B or 2C. If the interference amount is large, some signals may 
deviate to a numerical value region of the neighboring signal level. 
In the following description, a "predetermined numerical value region" 
means the region X shown in FIG. 4, i.e., a numerical value region having 
values larger than the maximum numerical value region G.sub.8 or smaller 
than the minimum numerical value region G.sub.1, and is clearly 
distinguished from the above-mentioned "numerical value regions". 
As described above, the slicer 4A shown in FIG. 3 obtains the estimated 
value w(n) {W.sub.i } of the received code value u(n), and outputs it to 
the subtracter 5 and an IIR digital filter 2. Also, the slicer 4A 
discriminates if the code value u(n) belongs to one of the predetermined 
numerical value regions, and outputs an identification signal D.sub.x that 
reflects this discrimination result. The subtracter 5 subtracts the sum 
y(n) output from an adder 3 from the estimated value w(n) output from the 
slicer 4A to calculate an error value e(n). The reception signal/error 
selection unit 6 selects one of the error value e(n) output from the 
subtracter 5 and a predetermined value, and supplies the selected value to 
a multiplier 7 connected at its output side. 
More specifically, when the code value u(n) belongs to none of the 
numerical value regions G.sub.1 to G.sub.8, the reception signal/error 
selection unit 6 outputs the error value e(n) supplied from the subtracter 
5 under the control of the identification signal D.sub.x that reflects 
this fact. Conversely, when the code value u(n) belongs to one of the 
numerical value regions G.sub.1 to G.sub.8, the reception signal/error 
selection unit 6 outputs a predetermined value under the control of the 
identification signal D.sub.x that reflects this fact. 
When the received code value u(n) belongs to one of the numerical value 
regions G.sub.1 to G.sub.8, for example, "zero" is set as the 
above-mentioned predetermined value to be output from the reception 
signal/error selection unit 6. This predetermined value need only express 
that the estimated value w(n) and the sum from the adder 3 match with each 
other, and may be appropriately determined in correspondence with 
calculation processing to be executed. 
The operation of the adaptive digital filter of this embodiment which 
comprises the above-mentioned slicer 4A and the reception signal/error 
selection unit 6 will be explained below. 
Referring to FIG. 3, a received code value u(n) is subjected to convolution 
calculation processing in the FIR digital filter 1, and a calculation 
result y.sub.1 (n) serves as one input value of the adder 3. 
The adder 3 receives a calculation result y.sub.2 (n) from the IIR digital 
filter 2 (to be described later) as the other input value, and adds it to 
the calculation result y.sub.1 (n) from the FIR digital filter 1. Then, 
the adder 3 supplies a sum y(n) to the slicer 4A and the subtracter 5. 
The sum y(n) serves as an equalization signal as the output of this digital 
filter. 
The slicer 4A discriminates if the sum y(n) input from the adder 3 belongs 
to one of the numerical value regions G.sub.1 to G.sub.8 or one of the 
predetermined value regions X shown in FIG. 4, as described above. When 
the sum y(n) belongs to one of the predetermined value regions X, the 
slicer 4A outputs a logic signal "1" as the identification signal D.sub.x 
to be supplied to the reception signal/error selection unit 6; when the 
sum y(n) does not belong to any of the predetermined numerical value 
regions X, it outputs a logic signal "0". 
On the other hand, when the slicer 4A determines that the sum y(n) output 
from the adder 3 belongs to one of the numerical value regions G.sub.1 to 
G.sub.8, it outputs the central value of the numerical value region to 
which the sum y(n) belongs to the subtracter 5 and the IIR digital filter 
2 as an estimated value w(n). Upon reception of the estimated value w(n), 
the IIR digital filter 2 performs convolution calculation processing of 
the input value, and supplies the calculation result y.sub.2 (n) to the 
adder 3. 
The reception signal/error selection unit 6 is controlled by the 
identification signal D.sub.x. That is, when the sum y(n) output from the 
adder 3 belongs to one of the predetermined value regions X, the selection 
unit 6 supplies the difference between the sum y(n) output from the adder 
3 and the estimated value w(n) output from the slicer 4A, i.e., the error 
value e(n) of the sum y(n) from the estimated value w(n), to the 
multiplier 7, and the multiplier 7 multiplies the input value with a 
convergence coefficient .alpha. to generate a coefficient sequence C.sub.k 
(n). 
The operation principle of the digital filter of this embodiment will be 
explained below. 
If the signal of a transmission code sequence is an 8 VSB code signal, the 
slicer 4A discriminates one of the numerical value regions G.sub.1 to 
G.sub.8 shown in FIG. 4 to which the received code value u(n) belongs, as 
described above. 
When the slicer 4A determines that the received code value u(n) belongs to, 
e.g., the numerical value region G.sub.7 shown in FIG. 4, the slicer 4A 
outputs the central value W.sub.7 of the numerical value region G.sub.7 as 
an estimated value w(n), and also outputs a logic signal "0" as the 
identification signal D.sub.x to the reception signal/error selection unit 
6. 
The reception signal/error selection unit 6 supplies the predetermined 
value "zero" as an error value to the multiplier 7 under the control of 
the identification signal D.sub.x. In this case, since the error value 
input to the multiplier 7 is zero, it is determined that the calculation 
result of a coefficient sequence has converged, and no calculation for 
updating the coefficient sequence is made, thus maintaining the previous 
coefficient sequence. 
Conversely, when the slicer 4A determines that the received code value u(n) 
has a signal level belonging to one of the regions X shown in FIG. 4, it 
outputs a logic signal "1" as the identification signal D.sub.x, and 
outputs one of signal levels W.sub.1 and W.sub.8 as a discrimination value 
w(n) of the input code value u(n). The reception signal/error selection 
unit 6 supplies the error value e(n) output from the subtracter 5 to the 
multiplier 7. The multiplier 7 makes a calculation for updating the 
coefficient sequence on the basis of the input error value e(n). 
In this case, as described above, the slicer 4A forcibly determines and 
outputs one of the central values W.sub.1 and W.sub.8 of the numerical 
value regions G.sub.1 and G.sub.8 as an estimated value w(n) of the code 
value u(n) included in one of the predetermined numerical value regions X. 
Since most of code values u(n) included in the predetermined numerical 
value regions X are those belonging to the numerical value region G.sub.1 
or G.sub.8 if they are not disturbed, an estimated value w(n) can be 
obtained with high precision without causing any discrimination error. 
In this case, since the central value W.sub.1 or W.sub.8 is forcibly 
determined as the estimated value w(n) for the code value u(n), the 
direction of error, i.e., the positive or negative sign of the error value 
e(n) can be correctly obtained. If an input code value u(n) belonging to 
one of the predetermined numerical value regions X is the one that should 
belong to a numerical value region other than the numerical value region 
G.sub.1 or G.sub.8, e.g., the numerical value region G.sub.7, the 
estimated value w(n) for this input code value u(n) can be determined in a 
numerical value direction in which the numerical value region G.sub.7 to 
which the code value u(n) should originally belong is present. In this 
manner, correct determination of the direction of error leads to an 
improvement in discrimination precision. 
As described above, according to the digital filter of the first 
embodiment, since a coefficient sequence is calculated using only code 
values belonging to the predetermined numerical value regions X shown in 
FIG. 4, a code sequence can be updated using codes about 1/8 the entire 
code sequence. For example, a signal of the ATV (Advanced TeleVision) 
system planned to be broadcast in U.S.A. includes a training sequence at a 
ratio of 1/300. With this embodiment, a coefficient sequence calculation 
can be executed using signal information about 40 times that of this 
system. Therefore, according to the digital filter of this embodiment, the 
calculation convergence time can be remarkably shortened. 
Since a coefficient sequence is updated using only a code signal belonging 
to the predetermined numerical value regions X shown in FIG. 4, a 
high-precision estimated value can be obtained. Therefore, as compared to 
a symbol decision method in which the calculation result diverges upon 
occurrence of the considerable intersymbol interference, the convergence 
precision can be greatly improved. 
Note that the digital filter of this embodiment can be constituted using a 
ROM (Read Only Memory). More specifically, as shown in FIG. 5, the slicer 
4A, the subtracter 5, the reception signal/error selection unit 6, and the 
multiplier 7 are replaced by a ROM 100 that receives as an address the sum 
y(n) output from the adder 3 shown in FIG. 3, and the calculation 
processing results of these slicer 4A, subtracter 5, reception 
signal/error selection unit 6, and multiplier 7 can be written in advance 
as tables in the ROM 100. 
In this digital filter, coefficient sequences corresponding to every 
calculation result of the adder 3 must be held in the ROM 100, and the 
correspondence between the calculation result of the adder 3 and the 
output from the ROM 100 must be determined in accordance with the 
above-mentioned LMSE algorithm. Note that any other type of storage may be 
used in place of the ROM 100 shown in FIG. 5 to realize the same 
arrangement, needless to say. 
(Second Embodiment) 
FIG. 6 is a block diagram showing the arrangement of an adaptive digital 
filter according to the second embodiment of the present invention. 
The digital filter of this embodiment has substantially the same 
arrangement as that of the digital filter of the first embodiment shown in 
FIG. 3, except that the digital filter of this embodiment comprises a 
subtracter 5A for calculating the difference between a training sequence 
Ref and an output signal y(n) from the adder 3, a training sequence/error 
selection unit 10 which is connected to the output side of the reception 
signal/error selection unit 6, selects the output from the subtracter 5A 
during the input period of the training sequence Ref, and selects the 
output from the reception signal/error selection unit 6 during other 
periods, and a center tap coefficient control unit 11 for controlling to 
attain a state in which the center tap coefficients of the FIR digital 
filter 1 and the IIR digital filter 2 are updated, during the input period 
of the training sequence Ref. 
FIGS. 7A to 7C respectively show the detailed arrangements of the reception 
signal/error selection unit 6, the training sequence/error selection unit 
10, and the center tap coefficient control unit 11. As shown in FIG. 7A, 
the reception signal/error selection unit 6 is constituted by a movable 
contact 6a and stationary contacts 6b and 6c. The stationary contact 6b 
receives a logic signal "0", and the stationary contact 6c receives an 
error value e(n) from the subtracter 5. The movable contact 6a is 
controlled by an identification signal D.sub.x output from the slicer 4A, 
and selects one of the logic signal "0" and the error value e(n) and 
supplies it to the output terminal. 
The training sequence/error selection unit 10 and the center tap 
coefficient control unit 11 respectively shown in FIGS. 7B and 7C have 
arrangements similar to that of the selection unit 6. The training 
sequence/error selection unit 10 shown in FIG. 7B selects one of the 
output from the reception signal/error selection unit 6 and a training 
sequence error value e.sub.ref (n) under the control of a training 
sequence period identification signal Dref, and supplies the selected 
value to the output terminal. On the other hand, the center tap 
coefficient control unit 11 shown in FIG. 7C selects one of logic signals 
"1" and "0" under the control of the training sequence period 
identification signal Dref, and supplies the selected signal to the output 
terminal. 
The operation of the adaptive digital filter of this embodiment with the 
above-mentioned arrangement will be explained below. 
The digital filter of this embodiment shown in FIG. 6 performs 
substantially the same operations as the digital filter shown in FIG. 3, 
except that it updates the center tap coefficient of the coefficient 
sequence on the basis of a training sequence Ref included in advance in a 
received code sequence U(n). Hence, the updating operation of the center 
tap coefficient as the characteristic feature of the present invention 
will be mainly explained. 
In the description of the operation, assume that a code sequence received 
from the transmission path includes a training sequence Ref in advance, as 
described above, and the code value of a code that the training sequence 
Ref means, i.e., the signal level, is agreed upon between the transmitter 
and the receiver in advance. In the following description, the 
pre-arranged code value corresponding to the signal level of the training 
sequence Ref will be referred to as a "value of the training sequence". 
Referring to FIG. 6, upon reception of the training sequence Ref included 
in a code sequence, the subtracter 5A calculates the difference between a 
sum y(n) from the adder 3 as the output from this digital filter and the 
value of the training sequence, and outputs an error of the sum y(n) with 
respect to the value of the training sequence, i.e., a training sequence 
error value e.sub.ref (n). 
On the other hand, the training sequence/error selection unit 10 selects 
the training sequence error value e.sub.ref (n) output from the subtracter 
SA under the control of a training sequence period identification signal 
Dref, and supplies it to the multiplier 7. The multiplier 7 generates a 
coefficient sequence on the basis of this training sequence error value 
e.sub.ref (n). At this time, the center tap coefficient control unit 11 
outputs a logic signal "1" during the input period of the training 
sequence and outputs a logic signal "0" during other periods, under the 
control of the training sequence period identification signal Dref. 
The operation of the integrator 42A of the FIR digital filter 1 is 
controlled in accordance with the logic signal output from the center tap 
coefficient control unit 11. More specifically, the center tap coefficient 
is updated on the basis of the training sequence error value e.sub.ref (n) 
during only the input period of the training sequence Ref, and the 
updating operation of the center tap coefficient is inhibited during other 
periods. 
As described above, since the center tap coefficient is updated using the 
training sequence Ref included in a code sequence, the calculation 
convergence time of a coefficient sequence can be further shortened as 
compared to the case wherein a coefficient sequence is updated using an 
estimated value as in the digital filter of the first embodiment. 
(Third Embodiment) 
FIG. 8 is a block diagram showing the arrangement of an adaptive digital 
filter according to the third embodiment of the present invention. 
The digital filter of this embodiment has substantially the same 
arrangement as that of the digital filter of the first embodiment shown in 
FIG. 3, except that the digital filter of this embodiment comprises a 
subtracter 5A for calculating the difference between a training sequence 
Ref and an output signal y(n) from the adder 3, a second slicer 30 for 
outputting a logic signal on the basis of the output from the subtracter 
5A, a counter 31 for counting the number of "l"s or "0"s of logic signals 
output from the second slicer 30, and an identification signal selection 
unit 32 for selecting one of a predetermined logic signal or an 
identification signal output from the slicer 4A on the basis of the count 
result, and supplying the selected signal to the reception signal/error 
selection unit 6. 
The operation of the adaptive digital filter of this embodiment with the 
above-mentioned arrangement will be explained below. 
The digital filter of this embodiment shown in FIG. 8 performs 
substantially the same operations as in the digital filter shown in FIG. 
3, except that it discriminates the convergence/non-convergence of a 
training sequence Ref included in a received code sequence, and selects an 
error value used in the calculation process of a coefficient sequence on 
the basis of the discrimination result. 
The coefficient sequence generation operation based on the discrimination 
result of the convergence/non-convergence of a training sequence as the 
characteristic feature of the present invention will be explained in 
detail below. 
In the following description of this embodiment as well, assume that the 
code sequence received from the transmission path includes a training 
sequence Ref in advance, the pre-arranged code value corresponding to the 
signal level of a training sequence Ref will be referred to as a "value of 
the training sequence", as in the above embodiment. 
Referring to FIG. 8, upon reception of a training sequence Ref included in 
a code sequence, the subtracter 5A calculates the difference between a sum 
y(n) from the adder 3 as the output from this digital filter and the value 
of the training sequence, and outputs an error of the sum y(n) with 
respect to the value of the training sequence, i.e., a training sequence 
error value e.sub.ref (n). 
The second slicer 30 compares the training sequence error value eref(n) 
output from the subtracter 5A with a predetermined allowable error amount 
during only the input period of the training sequence Ref. When the 
training sequence error value e.sub.ref (n) is smaller than the 
predetermined allowable error amount, the slicer 30 outputs a logic signal 
"1"; otherwise, it outputs a logic signal "0". Note that this 
predetermined allowable error amount is preferably set to be a value 
smaller than the intersymbol distance of a transmission code sequence. 
The counter 31 receives the logic signal output from the second slicer 30, 
and counts one of the values "1" and "0" of the logic signal. The counter 
31 is set in one of the following two operation modes in accordance with 
the count result, so as to determine which of logic signals "1" and "0" is 
to be counted in the next count operation. 
More specifically, the first mode is the divergence mode in which the value 
of the training sequence Ref is determined to change from a divergent 
state to a convergent state. In this divergence mode, the counter 31 
counts logic values "1" output from the second slicer 30 to detect 
convergence of the value of the training sequence. Note that the initial 
state of the counter 31 is set to count "1" On the other hand, the second 
mode is the convergence mode in which the value of the training sequence 
Ref is determined to change from a convergent state to a divergent state. 
In this convergence mode, the counter 31 counts logic values "0" output 
from the second slicer 30 to detect divergence of the value of the 
training sequence. 
When the count value of logic signals "1" of the counter 31 has exceeded a 
predetermined value in the divergence mode, the counter 31 determines that 
the training sequence Ref is in the convergent state, and outputs an 
operation inhibition signal to the identification signal selection unit 
32. Upon reception of the operation inhibition signal, the identification 
signal selection unit 32 supplies a logic signal "1" to the reception 
signal/error selection unit 6. In this case, the reception signal/error 
selection unit 6 supplies an error value e(n) output from the subtracter 5 
to the multiplier 7 at its output side under the control of the logic 
signal "1" as an identification signal. 
Thereafter, even after an elapse of the input period of the training 
sequence Ref, the reception signal/error selection unit 6 supplies the 
error value e(n) output from the subtracter 5 to the multiplier 7, and the 
multiplier 7 performs an updating calculation of a coefficient sequence on 
the basis of the error value e(n), as described above. 
When the next input period of the training sequence Ref has been reached, 
the operation mode is similarly discriminated. In this case, if the 
convergence mode was determined in the discrimination operation of the 
previous training sequence input period, the counter 31 counts logic 
values "0" output from the second slicer 30 during the current training 
sequence input period, so as to detect divergence of the training sequence 
Ref. 
When the count value of logic signals "0" output from the second slicer 30 
has reached a predetermined value, the counter 31 determines that the 
training sequence Ref is in the divergent state, and outputs an operation 
cancel signal to the identification signal selection unit 32. Upon 
reception of the operation cancel signal, the identification signal 
selection unit 32 supplies an identification signal D.sub.x output from 
the slicer 4A to the reception signal/error selection unit 6. In this 
case, the reception signal/error selection unit 6 supplies one of a 
predetermined value (e.g., zero) and an error value e(n) output from the 
subtracter 5 to the multiplier 7 at its output side under the control of 
the identification signal D.sub.x supplied from the slicer 4A. 
When the counter 31 counts logic values "1" or "0" from the second slicer, 
the previous count result must be deleted for each input period of the 
training sequence Ref. For example, when a channel is switched, and a 
large training sequence error value that considerably exceeds the 
allowable error amount is input to the second slicer 30, the second slicer 
30 may output a logic signal "0". In this case, the divergence mode is 
undesirably set based on a factor other than the training sequence. For 
this reason, once the output from the counter 31 is determined during the 
input period of a certain training sequence independently of the 
convergence or divergence mode, the determined state must be forcibly held 
until the next training sequence is input. 
As described above, according to the digital filter of this embodiment, 
since the convergence/non-convergence of a training sequence included in a 
received code sequence is monitored, the updating calculation of a 
coefficient sequence is performed using only received codes which are in a 
convergent state to some extent, and after the calculation of this 
coefficient sequence converges to some extent, the coefficient sequence is 
updated using all the received codes, thus further shortening the 
convergence time. 
(Fourth Embodiment) 
FIG. 9 is a block diagram showing the arrangement of an adaptive digital 
filter according to the fourth embodiment of the present invention. 
The digital filter of this embodiment is constituted by adding the second 
slicer 30, the counter 31, and the identification signal selection unit 32 
which constitute the digital filter shown in FIG. 8 to the digital filter 
shown in FIG. 6. 
More specifically, the digital filter of this embodiment shown in FIG. 9 
has substantially the same arrangement as that of the digital filter shown 
in FIG. 6, except that it additionally comprises a second slicer 30 for 
outputting a logic signal on the basis of the output from the subtracter 
5A, a counter 31 for counting the number of logic values "1" or "0" of 
logic signals output from the second slicer 30, and an identification 
signal selection unit 32 for selecting one of an identification signal 
D.sub.x output from the slicer 4A and a predetermined logic signal on the 
basis of the count result, and supplying the selected signal to the 
reception signal/error selection unit 6. 
The adaptive digital filter of this embodiment with this arrangement has 
the operations of both the second and third embodiments described above. 
More specifically, the digital filter of this embodiment updates the 
center tap coefficient of a coefficient sequence on the basis of a 
training sequence Ref included in a received code sequence U(n). In 
addition, the digital filter of this embodiment discriminates the 
convergence/non-convergence of the training sequence Ref included in the 
received code sequence during the input period of the training sequence 
Ref, and selects an error value to be used in the calculation process of a 
coefficient sequence on the basis of the discrimination result. 
Therefore, according to the digital filter of this embodiment, since the 
center tap coefficient is updated using the training sequence Ref, and the 
updating calculation of a coefficient sequence is performed using only 
received codes which are in a convergent state to some extent by 
monitoring the convergence/non-convergence of the training sequence Ref, a 
coefficient updating calculation is performed using all the received codes 
after the coefficient sequence calculation converge to some extent. For 
this reason, the calculation convergence time of a coefficient sequence 
can be further shortened as compared to the digital filter of the second 
or third embodiment. 
The above-mentioned digital filter of each of the first to fourth 
embodiments is constituted using both FIR and IIR digital filters, but may 
be constituted using either one of these filters. In this case, the adder 
3 shown in FIGS. 3, 5, 6, 8, or 9 may be omitted, and the calculation 
result of the FIR or IIR digital filter may be input to the slicer 4A and 
the subtracter 5. 
As can be seen from the above description, according to the present 
invention, the following effects can be obtained. 
According to the inventions of the first and third aspects, since the 
coefficient sequence is updated using a code value that belongs to the 
predetermined numerical value region other than the numerical value 
regions which are agreed upon in advance in correspondence with the code 
values, the estimation precision of a code can be remarkably improved, and 
the convergence time of the coefficient sequence calculation can be 
shortened. 
According to the inventions of the second and fourth aspects, since the 
adaptive digital filter comprises both a finite impulse response digital 
filter and an infinite impulse response digital filter, an increase in 
linear noise caused during the calculation process of the coefficient 
sequence can be suppressed in addition to the effect of the inventions of 
the first and third aspects. 
According to the invention of the fifth aspect, since the center tap 
coefficient is updated using a training sequence, the convergence 
precision of the coefficient sequence can be further improved, and the 
convergence time of the coefficient sequence calculation can be further 
shortened. 
According to the invention of the sixth aspect, since the adaptive digital 
filter comprises both a finite impulse response digital filter and an 
infinite impulse response digital filter, an increase in linear noise 
caused during the calculation process of the coefficient sequence can be 
suppressed in addition to the effect of the invention of the fifth aspect. 
According to the invention of the seventh aspect, since the convergence 
state of the reception signal is discriminated on the basis of the 
training sequence included in advance in the reception signal, calculation 
processing can be performed using only highly convergent codes, and the 
convergence time of the coefficient sequence calculation can be further 
shortened. 
According to the invention of the eighth aspect, both the effects of the 
inventions of the fifth and seventh aspects can be obtained. Furthermore, 
the convergence precision of the coefficient sequence can be further 
improved, and the convergence time of the coefficient sequence calculation 
can be further shortened. 
According to the inventions of the ninth and 10th aspects, since the 
predetermined numerical value region other than the numerical value 
regions that are agreed upon in advance in correspondence with the code 
values is set to be a numerical value region with a value larger than the 
value of the numerical value region whose absolute value corresponds to 
the maximum code value, the coefficient updating calculation can be 
performed using a larger number of code values than in the conventional 
training sequence method that utilizes a training sequence included at a 
very small ratio in the total transmission signal, thus greatly shortening 
the convergence time of the coefficient sequence calculation. 
According to the invention of the 11th aspect, since the coefficient 
sequence is calculated to minimize the error value, the coefficient 
sequence can be generated to remove disturbance signal components included 
in a code sequence.