JPH09270671A - Digital filter system - Google Patents

Abstract

PROBLEM TO BE SOLVED: To eliminate a noise component by arranging an adaptive filter to an intermediate frequency stage. SOLUTION: An input analog signal is sampled by a sampling frequency fs by a multiple of 4 of a carrier frequency fc. A demultiplexer 11 demultiplexes received data sequentially into four systems as four zero carrier signals whose frequency and amplitude are equal to each other but whose phases only differ from each other. The zero carrier signals are delayed by delay circuits 12-15 and the results are fed to adaptive filters 16-19. A multiplexer 22 multiplexes sequentially output data from the four adaptive filters 16-19 into one signal. A difference between the 1st zero carrier signal and the output of the 1st adaptive filter 16 is obtained to select a tap coefficient of the adaptive filters 16-19 based on the difference according to the LSM algorithm.

Description

Detailed Description of the Invention

[0001]

BACKGROUND OF THE INVENTION 1. Field of the Invention The present invention relates to a digital filter system for removing unnecessary signal components in an intermediate frequency stage.

[0002]

2. Description of the Related Art Conventionally, a filter for removing an unwanted wave in the field of wireless communication devices has been a high frequency stage (hereinafter referred to as RF).
Stage), an intermediate frequency stage (hereinafter, IF stage), and an audible frequency stage (AF stage).

For example, the applicant of the present application also sets a coefficient of a digital filter by using an adaptive algorithm to remove an unwanted wave in the AF stage and output only a target signal (“adaptive filter system”). Japanese Patent Application No. 6-237455, US Patent Application No. 0
8/452, 282). In addition, a noise canceling system for removing a large number of unnecessary waves has been proposed.

FIG. 9 shows an example of an adaptive filter arranged in the AF stage. The adaptive filter of FIG. 9 delays the AF input signal by the delay circuit 91 and supplies the delayed signal to the filter 92.
Output. The subtracter 93 obtains the difference between the AF input and the AF output. The control unit 94 uses the LMS (L
east Mean Square) filter according to algorithm 92
Control the characteristics of.

Further, for example, in a super-heterodyne receiver, an IF stage band pass filter (BP) is used.
F) is provided, and its pass band width is 7 to 15 KHz for both sideband waves (DSB) for broadcasting and the like, 2 to 4 KHz for one sideband wave (SSB), and telegraph (CW) Unwanted waves are removed by limiting the frequency to about 0.5 to 3 KHz.

However, in an amateur radio or a commercial radio capable of transmitting at an arbitrary frequency within an allowable range, there is a possibility that other radio waves (unwanted waves) may be mixed in the reception band (the pass band of the band pass filter). Not low. Further, since there are problems such as image interference, high frequency interference, and spurious beat mixing, a notch filter has been conventionally used to remove unnecessary waves in the pass band.

In recent years, an adaptive filter that uses a digital notch filter using digital signal processing, or an adaptive wave filter that uses a filter coefficient of a digital filter to attenuate only unnecessary wave signals and remove unnecessary waves in the reception band is used. In some cases, a digital notch filter system (hereinafter, simply notch filter system) or the like is used.

A notch filter system using other adaptive filters is disclosed in, for example, US Pat. No. 5,22.
There is one disclosed in 6,057. This notch filter system converts a received signal to an intermediate frequency and then passes the signal through a plurality of notch filters connected in series, while removing unnecessary waves, demodulating and outputting.

[0009]

The noise canceling system (adaptive filter system) previously filed by the present applicant is arranged in the AF stage. That is, it is arranged at the final stage of the receiver. Therefore, when circuits before the noise canceling system, for example, a demodulator and the like are realized by a digital signal processing circuit (DSP), a signal in which an unwanted wave is mixed is calculated, and thus the dynamic range of these circuits is increased. There is a problem of narrowing,
There is also a problem that the calculation error becomes large. Also, I
When feedback processing such as AGC (Automatic Gain Control) is performed in the F stage, processing such as AGC is performed based on the processed signal in which the unwanted wave remains mixed. Up to adjustment (suppression)
There is also the problem that it will be done. In particular, in the AGC process, when there is an unnecessary wave larger than the level of the target signal in the pass band, the target signal is gain-adjusted (suppressed) by the unnecessary wave, which causes a serious problem.
There is also a problem that it causes a decrease in (Signal-to-noise ratio). Therefore, it is desirable to remove unnecessary waves at least in the IF stage.

Further, when the notch filter system is realized by using digital signal processing, if the notch filters are connected in series as in the conventional case, the amount of operation is the number of operations in each digital filter connected. In addition to this, the amount of calculation of the LMS algorithm of each adaptive filter also increases. Therefore, the amount of calculation becomes very large. Further, as the amount of calculation increases, the calculation error may reach a level that cannot be ignored. Further, the possibility of oscillation (limit cycle) increases due to these arithmetic errors.

The present invention has been made in view of the above situation, and an object of the present invention is to provide a digital filter system capable of removing unnecessary components in a signal at an intermediate frequency stage.
Further, according to the present invention, unnecessary waves in the signal, beat interference, DC components from the A / D converter, etc. can be removed at the intermediate frequency stage,
Moreover, it is an object of the present invention to provide a digital filter system with a small amount of calculation in digital signal processing.

[0012]

In order to achieve the above object, a digital filter system according to a first aspect of the present invention is a digital filter system obtained by sampling an analog input signal at a sampling frequency N times the carrier frequency. Demultiplexer means for sequentially outputting signals to N systems, and N adaptive filters arranged at each of the N system outputs outputted from the demultiplexer means,
A control unit that controls the characteristics of the N adaptive filters based on at least one of the outputs of the N adaptive filters, and one output by sequentially selecting and outputting the outputs of the N adaptive filters. It is characterized by comprising multiplexer means for outputting as a signal and a multiplexer means.

The output of the demultiplexer means may be delayed and supplied to the N adaptive filters.

In the digital filter system according to the second aspect of the present invention, the digital signal obtained by sampling the analog input signal at the sampling frequency N times the carrier frequency is sequentially output to the N system. , At least one of conversion means for converting into N signals whose carrier frequency is substantially 0, N adaptive filters whose characteristics change according to an external signal, and output signals of the N adaptive filters. Based on the above, the control means controls the characteristics of the N adaptive filters, and the superimposing means that superimposes the output signals of the N adaptive filters and outputs the superposed signals as one signal. .

The output of the converting means may be delayed and supplied to the N adaptive filters. The control means may control the characteristics of the N adaptive filters based on the output of one of the N adaptive filters, for example.

The control means may control the characteristics of the N adaptive filters on the basis of all outputs of the N adaptive filters. In this case, the taps of the N adaptive filters may be divided into N groups, and the corresponding tap coefficient groups of the N adaptive filters may be controlled based on the outputs of the N adaptive filters.

A / D conversion means for sampling an analog input signal having a carrier frequency f _c at a sampling frequency f _s that is N times the carrier frequency, converting it into a digital signal, and supplying it to the demultiplexer means may be further provided. .

[0018]

BEST MODE FOR CARRYING OUT THE INVENTION A filter system according to an embodiment of the present invention will be described below. FIG. 1 shows the configuration of a filter system according to this embodiment. As shown, this filter system includes a demultiplexer (DMUX) 11 and first to fourth delay circuits (DELA).
Y) 12 to 15 and the first to fourth adaptive filters (FI
R) 16 to 19, a subtracter 20, and a control unit (LMS) 2
1 and a multiplexer (MUX) 22.

The demultiplexer 11 sequentially outputs each signal piece (data) of the intermediate frequency digital input signal S (t) as digital signals S ₁ (t) to S ₄ (t) from the first to fourth output ends. Output. The digital input signal S (t) is converted from analog to digital by sampling an intermediate frequency analog signal having a carrier frequency f _c (for example, 15.625 KHz) with a sampling clock having a frequency f _{s that is} four times f _c. This is the signal obtained by

The first to fourth delay circuits 12 to 15 delay the digital signals S ₁ (t) to S ₄ (t) output from the first to fourth output terminals of the demultiplexer 11 for a predetermined time. . Each of the first to fourth adaptive filters 16 to 19 is a filter that has a plurality of taps and that changes the passband width and the intermediate frequency of the passband according to the coefficient (tap coefficient) given to the taps by the control unit 21. , Digital signals S ₁ (t) to S output from the first to fourth delay circuits 12 to 15
Attenuating unnecessary components of ₄ (t), digital signal S ₁ '(t) ~ S
₄ '(t) is output.

The subtracter 20 controls the characteristics of the first to fourth adaptive filters 16 to 19 from the digital output signal S ₁ (t) at the first output terminal of the demultiplexer 11 to the first adaptive filter. The output S ₁ '(t) of the filter 16 is subtracted. Based on the output of the subtractor 20, the control unit 21 uses the well-known LMS.
According to the (Least Mean Square) algorithm, each adaptive filter 16 to 19 is arranged so that noise can be appropriately removed.
Control the tap coefficient of.

The multiplexer 22 receives the digital signal S ₁ '(t) that has passed through the first to fourth adaptive filters 16 to 19.
~ Select each data (signal piece) of S ₄ '(t) in order and
It outputs as one digital intermediate frequency signal (data string).

Next, the operation of the filter system having the configuration shown in FIG. 1 will be described. The IF digital input signal S (t) supplied to the demultiplexer 11 can be expressed by Equation 1. Here, x _i represents the i-th data of the digital input signal.

[0024]

## EQU1 ## S (t) = x ₁ , x ₂ , x ₃ , x ₄ . . . . x _i . . .

The demultiplexer 11 sequentially outputs the signal S (t) to the first to fourth output terminals. Therefore, the data x ₁ , x ₅ , x _{9 ,} . . . Are sequentially output, and the data x ₂ , x ₆ , x ₁₀ . . . Are sequentially output, and the data x ₃ , x ₇ , x are output from the third output end.
₁₁ ． . . Are sequentially output, and data x is output from the fourth output end.
₄ , x ₈ , x ₁₂ . . . Are sequentially output. Therefore,
The fourth output signals S ₁ (t), S ₂ (t), S ₃ (t), and S ₄ (t) are represented by equations 2-5.

[0026]

## EQU2 ## S ₁ (t) = x ₁ , x ₅ , x ₉ . . . .

## EQU3 ## S ₂ (t) = x ₂ , x ₆ , x ₁₀ . . . .

## EQU00004 ## S ₃ (t) = x ₃ , x ₇ , x ₁₁ . . . .

## EQU00005 ## S ₄ (t) = x ₄ , x ₈ , x ₁₂ . . . .

The sampling frequency f _s is four times the carrier frequency f _c of the analog input signal, and
It outputs in parallel to the grid. Therefore, the first to fourth signals S ₁ (t) to S ₄ (t) are sampling signals in the same phase as far as the carrier is concerned. Therefore, the first to the fourth
The signals S ₁ (t) to S ₄ (t) of 1 are so-called zero carrier signals with zero carrier, and are substantially AF band signals.

For example, when the IF digital input signal S (t) is the SSB signal represented by the equation 6, the first to fourth signals S
₁ (t) to S ₄ (t) are represented by Expressions 7 to 10. Number 7 to number 1
Signals S ₁ (t) to S ₄ (t) represented by 0 are signals in the AF band g
(T) and its Hilbert-transformed signal ^ g (t), and it can be seen that this is an AF band signal.

[0029]

[6] S (t) = g (t ) · cos (2 · π · f c · i /
_{f s) + ^ g (t} ) · sin (2 · π · f c · i / f s)

## EQU7 ## S ₁ (t) = {^ g (1), ^ g (5), ^ g (9), ...}

S ₂ (t) = {− g (2), −g (6), −g (10), ...}

[Equation 9] S ₃ (t) = {− ^ g (3), − ^ g (7), − ^ g (11), ...}

## EQU10 ## S ₄ (t) = {g (4), g (8), g (12), ...}

The first to fourth signals S ₁ (t) to S ₄ (t) are signals having the same frequency, the same amplitude, and different phases only. The first to fourth delay circuits 12 to 15 delay the signals S ₁ (t) to S ₄ (t) for a predetermined time in order to remove correlated noise.

The first to fourth adaptive filters 16 to 19 are
The unnecessary components of the input first to fourth signals are removed and output.

Since the input signal is a zero carrier signal,
The first to fourth adaptive filters 16 to 19 are digital filters having a pass band in the AF band, attenuate the unnecessary signal, and pass the target signal in the AF band. The characteristics of the first to fourth adaptive filters 16 to 19 are determined by the tap coefficient set by the control unit 21. Since the same tap coefficient is set by the control unit 21 in the first to fourth adaptive filters 16 to 19, they have the same characteristics.

The subtractor 20 obtains the difference between the output signal S ₁ (t) from the first output terminal of the demultiplexer 11 and the output signal of the first adaptive filter 16, and the control unit 21 is well known. Based on the LMS algorithm, the first to fourth adaptive filters 16 to 1 according to the signal supplied from the subtractor 20.
Control (correct) the tap coefficient of 9. Therefore, the characteristics of the first to fourth adaptive filters 16 to 19 are dynamically controlled according to the noise condition of the input signal, and the unnecessary components of the input signal can be appropriately removed.

The multiplexer 22 sequentially selects the output data of the first to fourth adaptive filters 16 to 19 and outputs it as a series of signals. That is, the first adaptive filter 16
The data (signal piece) output by X ₁ ', X ₅ ' ,. . . ,
The data output from the second adaptive filter 17 is X ₂ ',
X ₆ ',. . . , X ₃ ′, X ₇ ′,. . . , Fourth adaptive filter 19
The data output by X ₄ ', X ₈ ' ,. . . , And
The multiplexer 22 sequentially selects each data, X ₁ ',
_{X 2 ', X 3',} X 4 ', X 5', X 6 ',. . . , Is output. The signal composed of the data output from the multiplexer 22, that is, the signal after the noise is removed is
The carrier is the reproduced IF signal.

As described above, according to this embodiment, noise can be removed by using the adaptive filters 16 to 19 in the IF stage. Moreover, the adaptive filter 16-
As the reference numeral 19, one for the AF band can be used. As is generally known, an adaptive filter is more difficult to manufacture as it is used for a higher frequency band, and cost performance is lowered. However, according to this embodiment, since an adaptive filter for an AF band can be used, noise is reduced. Can be properly removed. Further, the cost can be suppressed.

The demultiplexers 11 to 22 do not have to be composed of individual parts, and they can be composed of a digital signal processor (DSP). In this case, demultiplexer 1
1 to multiplexer 22 is configured by each module that executes digital signal processing in the DSP.

In this embodiment, the band pass filter is used as the adaptive filter. However, for example, only a signal in a relatively narrow predetermined frequency band is attenuated,
It is also possible to use a notch filter that passes other wide band signals.

FIG. 2 shows the circuit configuration when a notch filter is used as the adaptive filter. The same parts as those in FIG. 1 are denoted by the same reference numerals. As shown in the drawing, in this case, the first to fourth output signals S ₁ (t) to S ₄ (t) from the demultiplexer 11 and the first to fourth adaptive notch filters 3
The difference between the output signals S ₁ '(t) to S ₄ ' (t) of
The difference data is calculated by the fourth subtractors 35 to 38 and input to the multiplexer 22. In addition, the control unit 21
The tap coefficients of the adaptive notch filters 31 to 34 are controlled according to the LMS algorithm on the basis of the output of the subtractor.

Also in the case of such a configuration, noise can be appropriately removed by using an adaptive filter in the IF stage.

1 and 2 show the present invention in TDAPF.
Although the example applied to (Time Division Adaptive Filter) is shown, the present invention is not limited to this, and TD-CDAPF (Time Division-Coeffici) is used.
It is also applicable to the ents Division Adaptive Filter (time division-coefficient division adaptive filter).

FIG. 3 shows TD-CDA as an adaptive filter.
An example in which PF is adopted is shown. In this configuration, the first to fourth adaptive filters 41 to 4 formed of bandpass filters are used.
The tap coefficient of 4 is divided into four groups H1 to H4. First output signal S ₁ (t) from the demultiplexer 11
And the output signal S ₁ '(t) of the first adaptive filter 41 is obtained by the subtractor 45. Based on this difference, the control unit 51 controls the first tap coefficient group H1 of the first to fourth adaptive filters 41 to 44 according to the LMS algorithm. Second output signal S from demultiplexer 11
The difference between ₂ (t) and the output signal S ₂ '(t) of the second adaptive filter 42 is obtained by the subtractor 46. Based on this difference, the control unit 52 follows the LMS algorithm and the second tap coefficient group H2 of the first to fourth adaptive filters 41 to 44.
Control.

The third output signal S ₃ (t) from the demultiplexer 11 and the output signal S ₁ '(t) of the third adaptive filter 43.
The difference between and is obtained by the subtractor 47. The control unit 53
Based on this difference, according to the LMS algorithm, the first
~ Controls the third tap coefficient group H3 of the fourth adaptive filters 41 to 44. The difference between the output signal S ₄ '(t) of the fourth output signal S ₄ (t) and the fourth adaptive filter 44 from the demultiplexer 11 is obtained by the subtracter 48. Based on this difference, the control unit 54 controls the fourth tap coefficient group H4 of the first to fourth adaptive filters 41 to 44 according to the LMS algorithm.

With such a configuration, the filter system can be constructed with a smaller processing amount than in the case where the TDAPF shown in FIGS. 1 and 2 is used. For example, a filter system may be a digital signal processor (TMS3
20C25) and the number of taps is 16, A
For the F band, 11.25 MIPS (Million Instructio
ns Per Second) and almost the same amount of calculation is required when TDAPF is used, but TD-CDAPF
Can be realized with 6.75 MIPS.

Next, as an application example of the filter system having the above configuration, an example of a radio receiver using these filter systems will be described with reference to FIG. This radio receiver includes an antenna 61 and an R received by the antenna 61.
F (high frequency) signal is selectively demodulated to intermediate frequency (IF)
A tuning / frequency converter 62 for converting into a signal, an A / D converter 63 for A / D converting the IF signal output from the tuning / frequency converter 62, and noise included in the output of the A / D converter 63. , A detection circuit 65 for reproducing and outputting an AF signal from an output signal (IF signal) of the filter system 64, and a D / A conversion for D / A converting and outputting the reproduced AF signal. Device 66, an amplifier 67 that amplifies the analog audio signal output from the D / A converter 66, and a speaker 68 that is driven by the output of the amplifier 67 and emits sound.

In the wireless receiver, it is possible to remove the noise contained in the received signal at the AF stage. But,
When the filter system is arranged in the AF stage, noise may be mixed in the processed signal in the IF stage, the dynamic range of these circuits may be narrowed, and the calculation error may be increased. Also, AGC
Undesired waves are mixed in the processing signal of feedback processing such as
There is also a problem that the signal level of the received signal is adjusted (suppressed) by the processed signal mixed with the unwanted wave. According to the configuration of FIG. 4, noise is removed by the IF stage filter system 64, and thereafter, detection, D / A conversion, amplification, etc. are performed, so that data processing that is less affected by noise becomes possible.

Computer simulations were performed for the conventional filter system for processing the AF band and the filter system for processing the IF band composed of TDAPF and TD-CDAPF to compare the convergence characteristics.

[Simulation Condition] For comparison between the AF band filter system and the IF band filter system, the model shown in FIG. 5 is used for the AF band and the model shown in FIG. 6 is used for the IF band. use.

That is, in the simulation model for the AF band, as shown in FIG. 5, the AF digital input signal is modulated by the SSB modulator (MOD) 71 into an IF signal. Then, this IF signal is sent to the SSB demodulator (DEMO
D) Demodulate by 72 to obtain an AF signal. This AF signal is A / D converted and supplied to a conventional AF filter system 73 shown in FIG. 9 to obtain an AF digital output signal and an error signal.

On the other hand, in the simulation model for the IF band, as shown in FIG. 6, the AF digital input signal is modulated into the IF signal by the SSB modulator (MOD) 81. Subsequently, the IF signal is A / D converted and supplied to the IF filter system 82 having the configuration shown in FIG. 1 or 3.
The output of the IF filter system 82 is the SSB demodulator (DE
MOD) 83 to demodulate and obtain an AF digital output signal. Since the error signal output from the IF filter system 82 is also an IF signal, it is supplied to the SSB demodulator 84 to obtain a demodulated signal of the error signal.

The input signal is a tone signal of 1 KHz and 2.4 KHz to which white noise is added, and SNR (Signal
Simulation is performed for two cases of -to-Noise Ratio) of 24 dB and 10 dB. The variable err (t) defined by Equation 11 was used for evaluation of the convergence property.

[0051]

Err (t) = E [20 · log ₁₀ | Er1 (i) |] (t ≦ i ≦ t + 500)

Here, E [•] represents an expected value operation, and in this simulation, i represents an expected value from (t) to (t + 500). The error signal is Er1 for the AF band, but the error signal is IF1 for the IF band.
Since it is a signal, the demodulated error signal is Er1. The values shown in Table 1 were used as the number of taps, step parameters, and delay time of the delay circuit of each adaptive filter constituting the AF band noise filter system and the IF band noise filter system.

[0053]

[Table 1] AF: AF band filter system TDAPF: IF band TDAPF filter system TD-CDAPF: IF band TD-CDAPF filter system

[Simulation Results] Under these conditions, the variable err (t) when the SNR of the input signal is 24 dB is shown in FIG. 7, and the variable err (t) when the SNR is 10 dB is shown in FIG. Shown in. From FIGS. 7 and 8, it is the TD-CDAPF filter system that converges the fastest. The TDAPF filter system is
When SNR = 24 dB, it shows substantially the same convergence characteristics as the TD-CDAPF filter system,
At SNR = 10 dB, the convergence is slower than that of the AF band filter system. From the above, TD-CDAP
It has been confirmed that the F-type filter system is the most effective method because it converges fastest and requires a small amount of processing.

The present invention is not limited to the above embodiment, and various modifications and applications are possible. For example, in the filter system shown in FIGS. 1 to 3, the sampling frequency fs is set to four times the carrier frequency fc of the analog input signal, and the digital signal after A / D conversion is divided into four systems by the demultiplexer. , Sampling frequency fs is N of carrier frequency fc of the analog input signal
(N is a positive positive number of 4 or more) times, and the digital signal after A / D conversion may be divided into N systems by a demultiplexer. Besides, the circuit configuration and the like can be arbitrarily changed.

[0056]

As described above, according to the present invention, noise can be dynamically reduced by using the adaptive filter in the intermediate frequency stage. Moreover, the adaptive filter itself may be in the audible frequency band, and a high performance filter can be easily used.

[Brief description of drawings]

FIG. 1 is a block diagram showing a configuration of a filter system according to an embodiment of the present invention.

FIG. 2 is a block diagram showing a configuration of a modified example of the filter system shown in FIG.

FIG. 3 is a block diagram showing a configuration of a modified example of the filter system shown in FIG.

FIG. 4 is a block diagram showing a configuration example of a receiver using the filter system shown in FIGS.

5 is a block diagram showing a configuration of a circuit used for measuring characteristics of the filter system shown in FIG. 9. FIG.

FIG. 6 is a block diagram showing a configuration of a circuit used to measure characteristics of the filter system shown in FIGS. 1 to 3.

FIG. 7 is a graph showing a simulation result of a convergence characteristic of the filter system in the circuits shown in FIGS. 5 and 6;

FIG. 8 is a graph showing a simulation result of a convergence characteristic of the filter system in the circuits shown in FIGS.

FIG. 9 is a block diagram showing a configuration of an adaptive filter arranged in a conventional AF stage.

[Explanation of symbols]

 11 Demultiplexer 12-15 Delay circuit 16-19 Adaptive filter 20 Subtractor 21 Control part 22 Multiplexer 31-34 Adaptive notch filter 35-38 Subtractor 41-44 Adaptive filter 45-48 Subtractor 51-54 Control part

Claims (5)

[Claims] 1. Demultiplexer means for sequentially outputting digital signals obtained by sampling an analog input signal at a sampling frequency N times as high as a carrier frequency, and N system outputs outputted from the demultiplexer means. Of N adaptive filters, control means for controlling characteristics of the N adaptive filters based on at least one of outputs of the N adaptive filters, and N adaptive filters of the N adaptive filters. A digital filter system comprising: multiplexer means for outputting a single signal by sequentially selecting and outputting the outputs. 2. The digital filter system according to claim 1, wherein the control means further comprises delay means for delaying the output of the demultiplexer means and supplying the delayed output to the N adaptive filters. 3. A digital signal obtained by sampling an analog input signal at a sampling frequency N times as high as the carrier frequency is sequentially output to N systems, whereby N signals having a carrier frequency of substantially 0 are obtained. The characteristics of the N adaptive filters are controlled based on at least one of conversion means for converting, N adaptive filters whose characteristics change according to an external signal, and output signals of the N adaptive filters. A digital filter system comprising: a control means; and a superposition means for superposing the output signals of the N adaptive filters and outputting the superposed signals as one signal. 4. The digital filter system according to claim 3, wherein the control means further comprises delay means for delaying the output of the converting means and supplying the delayed output to the N adaptive filters. 5. The N adaptive filters change their characteristics based on tap coefficients supplied from the outside, and the tap coefficients of each adaptive filter are logically divided into N groups. Controlling the corresponding tap coefficient group of the N adaptive filters based on each output of the N adaptive filters. Filter system.