JPH0764567A - Active noise controller - Google Patents

Abstract

PURPOSE:To obtain a silencing effect which is stable in a long term by performing an arithmetic processing to data obtained by converting analog signals detected by first and second mechano-electric transducer into digital signals by a specific digital filter. CONSTITUTION:An acoustic wave is detected by microphones M1 and M2, the output signal of the microphone M2 is inputted to a control algorithm 10 as a differential signal E, the output signal from the microphone M1 is added to the output signal of a filter 7 with opposite phases by an adding point 6, and inputted to filters 8 and 9. The output signal of the filter 9 is inputted to a control algorithm 10, a transfer function to be applied to a filter 8 is decided so that the differential signal E can be the minimum by the control algorithm 10, and a filter coefficient is applied to the filter 8, and converted into a prescribed phase characteristic signal by the filter 8. The output signal of the filter 8 is D/A converted, and outputted to a speaker S as an added sound source which emits the acoustic wave for silencing at the position of the microphone M2.

Description

Detailed Description of the Invention

[0001]

BACKGROUND OF THE INVENTION 1. Field of the Invention The present invention relates to an active noise control device, and more particularly to an active noise control device capable of performing stable active noise control over a long period of time and shortening a duct length.

[0002]

2. Description of the Related Art FIG.
FIG. 6 is a system diagram of a conventional active noise control device disclosed in Japanese Patent Publication No. In the figure, 30 is an acoustic system, which comprises an axially extending duct 32 having an input end 34 for receiving an input sound wave and an output end 36 for emitting an output sound wave. The sound wave that causes noise passes from left to right in the axial direction in the duct 32 that constitutes the propagation path of the sound wave. The adaptive filter model 38 includes a first electromechanical converter (hereinafter, referred to as “input converter”) including a microphone.
The signal detected at 42 is a high-pass filter 62 having a cutoff frequency of 4.5 hertz and a cutoff frequency of 500.
Input signal 40 passed through a Hertz low-pass filter 54
And a signal detected by a second electromechanical converter (hereinafter referred to as “error converter”) 46 composed of a microphone, and a high-pass filter 56 having a cutoff frequency of 45 hertz.
And the error signal 44 that has passed through the low-pass filter 58 having a cutoff frequency of 500 Hertz, and the error signal 4
The correction signal 48 modeled so that 4 approaches a predetermined value such as 0 is output. The correction signal 48 passes through a low-pass filter 60 and is input to a mechanical-electrical converter (hereinafter, referred to as “output converter”) 50 composed of an omnidirectional output speaker, and the canceling sound wave emitted from the output converter 50 is input. , Is guided into the duct 32 so as to attenuate the output sound wave.

FIG. 12 shows a second conventional acoustic system, and the same reference numerals as those in FIG. 11 denote the same or corresponding parts. As the input signal 40, a signal detected by the input converter 42, high-passed by the high-pass filter 101 having the cut-off frequency f1, and low-passed by the low-pass filter 106 having the cut-off frequency f6 is input. The error signal 44 is detected by the error converter 46, high-passed by the high-pass filter 102 having the cutoff frequency f2, further high-passed by the high-pass filter 103 having the cutoff frequency f3, and cutoff frequency f4.
The signal which is low-passed by the low-pass filter 104 and is further input by the low-pass filter 105 having the cutoff frequency f5 is input. In the second embodiment, as shown in FIG. 13, f1 <f2 <f3 <f4 <f5 <f6 is configured, and the high-pass filters 102 and 103 pass the error signal 44 in multiple stages in the high band, and further, the low-pass signal is passed. The filters 104 and 105 provide multistage low pass. This multiple pass shapes the filter response at the roll-off frequency. Further, the frequency band of the input signal 40 that has been passed through the high band and then the low band is
Error signal 4 that has been passed through the multi-stage high band and then the multi-stage low band
It is larger than the frequency band of 4.

Next, FIG. 14 and FIG. 15 are diagrams respectively showing the amplitude characteristic and the delay characteristic in the Butterworth type low-pass analog filter. In FIG. 14, the horizontal axis represents the normalized frequency, the vertical axis represents the amplitude, and the order is the parameter. In FIG. 15, the horizontal axis represents the normalized frequency, the vertical axis represents the delay, and the order is the parameter. From both characteristic diagrams, it can be seen that as the order increases, the cutoff frequency becomes steeper, the change in delay time is not constant, and the group delay rapidly increases near the cutoff.

16 and 17 show amplitude characteristics and delay characteristics of a Bessel type low-pass analog filter, respectively. In FIG. 16, the horizontal axis represents the normalized frequency, the vertical axis represents the amplitude, and the parameter is the order. In FIG. 17, the horizontal axis represents the normalized frequency, the vertical axis represents the delay, and the order is taken as a parameter. From both characteristic diagrams, it can be seen that the cutoff frequency becomes steeper as the order becomes higher, but it becomes looser than the Butterworth type. Also, the change in the delay time is smaller than that in the Butterworth type, but it is not yet constant, and it can be seen that the group delay increases near the cutoff.

It is clarified here that if the Butterworth type having a good blocking characteristic is selected, the group delay characteristic becomes poor, and if the Bessel type having a good group delay characteristic is selected, the blocking characteristic becomes poor. Furthermore, although the low-pass filter has been described here, the same applies to a high-pass filter and a band-pass filter.

[0007]

Since the conventional active noise control system is constructed as described above, the sensor microphone M1 and the speaker S as an additional sound source are used where the length of the duct needs to be shortened. It is necessary to shorten the distance, and considering the arithmetic processing time of the DSP, the delay time of the analog filter, the delay time of the speaker, etc., it cannot be so short.

Further, in order to operate stably, an external analog filter is required to remove an unnecessary input signal, and since it is composed of physical hardware, its price is unavoidable.

Further, if the analog filter has a strict specification for removing unnecessary signals, the input signal is always subjected to phase distortion at the cutoff point, and as a result, the correct input signal is not input to the DSP, so that the calculation result is There may be a case where a deviation occurs, the noise reduction effect is small, or the system becomes unstable and the system diverges, and the noise control cannot be performed.

The present invention has been made to solve the above problems, and an object of the present invention is to obtain an active noise control device which can shorten the duct length and obtain a stable silencing effect in the long term. .

[0011]

In the active noise control system according to the present invention, the analog signal detected by the first and second electromechanical converters is specified with respect to the data converted into the digital signal. It is provided with arithmetic processing means for performing arithmetic processing with a digital filter.

In the active noise control system according to the present invention, the transfer function of the FIR adaptive digital filter modified by the control algorithm is provided with a calculating means for convoluting the transfer function of the specific digital filter.

Further, the characteristic of the specific digital filter is a high pass filter, a band pass filter or a low pass filter.

The FI modified by the control algorithm
The R-adaptive digital filter is provided with a convolution calculation means for convoluting the transfer function of the digital filter and the transfer function of a specific digital filter, and a time shift means for time-shifting the transfer function.

Further, a calculating means for convoluting the transfer function of the specific digital filter with the transfer function of the FIR adaptive digital filter according to the result of the first judging means for judging that the amount corrected by the control algorithm has become small. Be prepared.

An operation of convoluting the transfer function of the specific digital filter with the transfer function of the FIR adaptive digital filter according to the result of the second judging means for judging the level of the output signal of the second electromechanical converter. It is equipped with means.

The time zone in which the transfer function of the specific digital filter is subjected to the convolution operation processing is a collection of time fields other than the time field processed by the control algorithm.

[0018]

The active noise control device configured as described above is
In order to remove the unnecessary input signal with a specific digital filter, the specifications of the external analog filter are minimized, so the scale of the hardware becomes smaller accordingly.
Further, the distortion of the phase of the input signal due to the analog filter is minimized, and noise control capable of obtaining a stable silencing effect in the long term can be performed.

Also, the total processing time from detecting noise to producing a result is shortened, and the length of the duct can be shortened to some extent.

[0020]

【Example】

Example 1. First Embodiment FIG. 1 is a system block diagram showing a first embodiment of the present invention. In the figure, two sensor microphones M1 and M2 for detecting a sound wave propagated from a noise source in a sound wave propagation path 1 are installed at positions upstream and downstream of the speaker S as an additional sound source. Has been done. The signal from the sensor microphone M1 passes through a preamplifier 2 for amplifying the signal and a low-pass filter 3 for anti-aliasing, and an analog signal is converted into a digital signal by an A / D converter 4, and a first FIR digital filter 5 is supplied. It is input to the addition point 6 via.

Further, the output signal of the acoustic feedback suppressing digital filter 7 is input, and the output signal of the digital filter 7 is added in an opposite phase to the output signal from the sensor microphone M1. In addition, addition point 6
Is configured to be input to the FIR adaptive digital filter 8 and the second FIR digital filter 9, and the output signal X of the second FIR digital filter 9 is input to the control algorithm 10. .

Further, in the control algorithm 10, the output signal of the sensor microphone M2 passes through the preamplifier 11 for amplifying the output signal and the low-pass filter 12 for anti-aliasing, and the analog signal is converted into a digital signal by the A / D converter 13. And is input as an error signal E through the third FIR digital filter 14.

The output signal of the FIR adaptive digital filter 8 becomes the input signal of the acoustic feedback suppressing digital filter 7 and the D / A converter 15, and D / A
The output of the A converter 15 is passed through the low pass filter 16 and the speaker driving power amplifier 17 to the speaker S.
Entered in.

In the above structure, the sound waves propagated from the noise source are detected by the sensor microphones M1 and M2,
The output signal of the sensor microphone M2 is the error signal E.
Is input to the control algorithm 10. At the addition point 6, the output signal from the sensor microphone M1 and the output signal of the acoustic feedback suppressing digital filter 7 are added in opposite phases to each other, and the addition output is the FIR adaptive digital filter 8 and the second FIR digital filter 9 Entered in. The output signal of the second FIR digital filter 9 is input to the control algorithm 10, and the control algorithm 10 outputs the output signal X and the error signal E of the second FIR digital filter 9 so that the error signal E becomes the minimum. Based on this, the transfer function to be applied to the FIR adaptive digital filter 8 is determined, and a filter coefficient, which is a control parameter for specifying the transfer function, is applied to the FIR adaptive digital filter 8.

In the FIR adaptive digital filter 8,
The input signal is converted into a signal having predetermined amplitude and phase characteristics based on the given filter coefficient. The output signal of the FIR adaptive digital filter 8 is D / A converted and output to the speaker S as an additional sound source that emits a sound wave for silencing in order to eliminate the sound wave propagated from the noise source at the position of the sensor microphone M2. . In this way, the sound wave propagated from the noise source is canceled at the position of the sensor microphone M2.

The sound wave for silencing from the speaker S is detected by the sensor microphone M1. This component is the output of the digital filter 7 that reproduces the transfer characteristic from the sound absorbing FIR adaptive digital filter 8 to the addition point 6. The signal is reversed in phase and the sensor microphone M1
Is canceled by adding the output signal of the sensor S and the output point of the sensor S to the sensor microphone M.
Acoustic feedback to 1 is suppressed. That is,
The digital filter 7 acts as a digital filter for suppressing acoustic feedback.

FIG. 2 is a diagram showing the configuration of the active noise control system shown in FIG. 2, the same reference numerals as those in FIG. 1 denote the same or corresponding parts, and 20 denotes a DSP.
(Digital signal processor), 21 is a display,
These are connected to each other via a bus line 22.

The digital signal processor 20 includes a first FIR type digital filter 5, a second FIR type digital filter 9, a third FIR type digital filter 14, a FIR adaptive type digital filter 8 and an acoustic feedback suppression shown in FIG. Digital filter 7,
The control processor 10 executes the role of the processing of the control algorithm 10 and also controls the overall system, transfers data to the A / D converters 4 and 13 and the D / A converter 15, and outputs data to the display unit 21. Is going.

Thus, the sensor microphone M
1 and M2, the output signal of the sensor microphone M2 is input to the control algorithm 10 as an error signal E. At the addition point 6, the output signal from the sensor microphone M1 and the output signal of the acoustic feedback suppressing digital filter 7 are added in opposite phases to each other, and the addition output is the FIR adaptive digital filter 8 and the second FIR digital filter 9 Entered in. The output signal of the second FIR type digital filter 9 is input to the control algorithm 10, which controls the second FIR so that the error signal E is minimized.
Type digital filter 9 output signal X and error signal E
The transfer function to be applied to the FIR adaptive digital filter 8 is determined based on the above, and a filter coefficient, which is a control parameter for specifying the transfer function, is applied to the FIR adaptive digital filter 8.

If the amplitude characteristics of the first FIR digital filter 5 and the third FIR digital filter 14 are the amplitude characteristics of the low-pass analog filter of FIG. 14, the delay characteristics are as shown in FIG. Become. In FIG. 3, the horizontal axis represents frequency and the vertical axis represents delay time. Even if the order of the low-pass analog filter becomes higher, the change in delay time is constant with respect to the change in frequency, so the group delay becomes constant, and it can be seen that there is no change at or near the cutoff point. .

As is well known and written in a technical book, the FIR type digital filter is basically characterized in that a group delay can be made constant, and the phase does not change. , There is no rapid phase change near the cutoff frequency. Therefore, the sensor microphone M
The signals detected by 1 and M2 can be truly input to the FIR adaptive digital filter 8 and the second FIR digital filter 9. Further, although the anti-aliasing low-pass filter is processed by an analog circuit, the frequency range processed by noise control is as shown in FIG.
It is needless to say that a more faithful noise control can be performed by configuring the digital filter so as to control in the portion B sufficiently lower than the cutoff frequency A of the analog filter as shown in FIG.

Example 2. Second Embodiment FIG. 5 is a system block diagram of an active noise control system showing a second embodiment of the present invention. In the figure, the same reference numerals as those in FIG. 1 indicate the same or corresponding portions, and the same processing is performed. Two sensor microphones M1 and M2 are provided in the sound wave propagation path 1.
And the speaker S are provided, and the sensor microphone M1
Signal passes through the preamplifier 2 and the low pass filter 3,
It is converted into a digital signal by the A / D converter 4 and input to the addition point 6. Further, the output signal of the acoustic feedback suppressing digital filter 7 is added to the addition point 6 in the opposite phase to the output signal from the sensor microphone M1, and the output signal of the addition point 6 is added to the FIR adaptive digital filter 8 And to the second FIR digital filter 9.

The output of the second FIR digital filter 9 is input to the control algorithm 10. Further, the output signal of the sensor microphone M2 as the error signal E passes through the preamplifier 11 and the low-pass filter 12, is converted into a digital signal by the A / D converter 13, and is input to the control algorithm 10. The fourth digital filter 18 performs a convolution operation with the transfer function of the FIR adaptive digital filter 8, and then outputs the output signal of the FIR adaptive digital filter 8 to the acoustic feedback suppressing digital filter 7 and the D / A converter 15.
And the output of the D / A converter 15 is a low-pass filter 16 and a speaker driving power amplifier 17
Is input to the speaker S via.

In the above configuration, the sound wave propagated from the noise source is detected by the sensor microphones M1 and M2, and the output signal of the sensor microphone M2 is input to the control algorithm 10 as the error signal E. At the addition point 6, the output signal from the sensor microphone M1 and the output signal from the acoustic feedback suppressing digital filter 7 are added in opposite phases to each other, and the addition output is F
It is input to the IR adaptive digital filter 8 and the second FIR digital filter 9. The output signal of the second FIR type digital filter 9 is the control algorithm 10
The control algorithm 10 inputs the transfer function to be given to the FIR adaptive digital filter 8 based on the output signal X of the second FIR digital filter 9 and the error signal E so that the error signal E becomes minimum. The FIR adaptive digital filter 8 is provided with a filter coefficient which is a control parameter for determining and determining the transfer function. After that, the transfer function of the fourth digital filter 18 and the transfer function of the FIR adaptive digital filter 8 are subjected to convolution calculation, and the final filter coefficient is given to the FIR adaptive digital filter 8.

Thus, the sensor microphone M
The signals of 1 and M2 pass only a low-pass filter for anti-aliasing of the A / D converter, and the signals become the input signals of the control algorithm. Therefore, the transfer function of the FIR adaptive digital filter 8 is It is generated as a transfer function in a wide frequency range, and in particular, a low-pass frequency component is often included for a long time, so that the system often becomes unstable. For example, the playback capacity of the speaker is 40
When the frequency of the component of 40 hertz or less is input and the signal of 5 hertz which does not have the reproduction ability comes in, the control algorithm is designed to reduce the noise of 5 hertz. The transfer function of the FIR adaptive digital filter 8 is generated. However, since the speaker has no reproduction capability, the frequency component of 5 Hertz does not actually drop. Even so, the control algorithm generates the transfer function of the FIR adaptive digital filter 8 so that the frequency component of 5 hertz falls further.
The transfer function of the adaptive digital filter 8 continues to grow,
Eventually, the system may become uncontrollable and may diverge.

However, if the transfer function of the fourth digital filter 18 is the transfer function shown in FIG. 6 in the frequency domain,
The response is reduced from the point C at 40 Hz where the reproduction capability is reduced, and is eliminated as a response at 5 Hz. Since this filter is always subjected to the convolution operation, the transfer function of the FIR adaptive digital filter 8 does not continue to grow and is stable. In order to obtain the required transfer function, it can be easily obtained with an existing digital filter design support tool. The block diagram of this active noise control device is the same as that of FIG.

Example 3. First and third described in the first embodiment
The phase characteristics of the digital filters 5 and 14 and the fourth digital filter 18 described in the second embodiment change linearly regardless of whether they are high-pass filters, low-pass filters, or band-pass filters due to the characteristics of FIR digital filters. However, since the group delay is constant, an appropriate filter can be selected and used according to the system.

Example 4. In the active noise control device described in the second embodiment, the FIR corrected by the control algorithm
When the transfer function of the adaptive digital filter is the impulse characteristic shown in FIG. 7, the specific digital filter is a low-pass filter, and the transfer function is the impulse characteristic shown in FIG. 8, the transfer functions of the two digital filters are shown. The result of the transfer function obtained by performing the convolution calculation of is the impulse characteristic shown in FIG. In this case, the main delay is from TA seconds in FIG. 7 to TB seconds in FIG. 9, and as it is, even if sound is emitted at the point of the speaker S, TB-TA seconds have already been delayed, so the expected noise reduction effect can be obtained. Absent. Therefore, the FIR whose delay time is modified by the control algorithm is
The expected silencing effect is obtained by shifting or circulating the time by TB-TA seconds so as to be the same as the delay time of the transfer function of the adaptive digital filter, and using the result as the final transfer function of the FIR adaptive digital filter. Is obtained.

Example 5. In the active noise control device described in the second or fourth embodiment, the control algorithm F
The transfer function of the IR adaptive digital filter is modified, but in the system that performs active noise control for the first time,
Unless the transfer function is measured beforehand by some device, the transfer function is completely unknown, so it starts from "zero". Therefore, it is meaningless to convolve the transfer function of the fourth digital filter before the transfer function has been completely generated, so that the control algorithm is FI.
The convolution calculation of the transfer function of the fourth digital filter is started when the amount of modification of the transfer function for the R adaptive digital filter becomes small.

Example 6. In the active noise control device described in the second or fourth embodiment, similarly to the fifth embodiment, in the system in which the active noise control is performed for the first time, the output signal of the second mechanical-electrical converter is, for example, 6 from the first level.
When the level becomes lower by dB, the convolution operation of the transfer function of the fourth digital filter with the transfer function of the FIR adaptive digital filter may be started.

Example 7. In the active noise control device described in the second, fourth, or fifth embodiment, all processing needs to be completed within one sample. For example,
If the sampling frequency of the A / D converter is 3 kilohertz, one sample takes about 333 microseconds, and a series of processing needs to be completed within this 333 microsecond period. However, although a DSP (digital signal processor) is usually used as a method for realizing a digital filter, the current state of the DSP is still limited even though the processing speed is fast, and the FIR adaptation modified by a control algorithm is still available. The processing time for convoluting the transfer function of the fourth digital filter with the transfer function of the digital filter is very large and takes a long time.

However, the transfer function of the fourth digital filter is convolved with the transfer function of the FIR adaptive digital filter modified by the control algorithm. Normally, the transfer function of the FIR adaptive digital filter is calculated. When the noise control is performed, the transfer function hardly changes and has the same impulse characteristic. That is, if the transfer function of a large system due to the acoustic space in the propagation path changes, it only follows the change, so the change is extremely small, and it hardly changes in a few seconds in most cases.

Therefore, the transfer function of the FIR adaptive digital filter modified by the control algorithm has the fourth
The convolution calculation processing of the transfer function of the digital filter may be completed within several samples instead of within one sample. However, since the processing of other digital filters must be completed within one sample, as shown in FIG. 10, the period of SA is set to one sample time and the period of SB is set to the digital filter required for normal noise control. When the time relating to the calculation is used, it is the time when the SC period is open. Therefore,
The transfer function of the fourth digital filter is convoluted in the SD period in which the SC period is collected. That is, the processing is performed in a time division manner.

[0044]

The phase change of the analog filter in the vicinity of the cutoff frequency is difficult to accurately represent with a model, but since the phase is basically not changed in the digital filter, it is in the vicinity of the cutoff frequency. The instability due to the phase change of is eliminated, and a long-term stable silencing effect can be obtained.

Further, an externally provided analog filter (high-pass filter) is unnecessary, and an analog low-pass filter having relatively loose characteristics can be used. Therefore, the number of parts of the analog filter section can be reduced and the cost can be reduced. it can.

Further, the delay time is shortened, and as a result, noise control in a short propagation path becomes possible.

[Brief description of drawings]

FIG. 1 is a system block diagram of an active noise control device according to a first embodiment of the present invention.

FIG. 2 is a diagram showing a configuration of an active noise control device according to a first embodiment and a second embodiment of the present invention.

FIG. 3 is a delay characteristic diagram of a digital filter.

FIG. 4 is a characteristic diagram showing a relationship between an analog low-pass filter and a digital low-pass filter according to the first embodiment.

FIG. 5 is a system block diagram of an active noise control system according to Example 2 of the present invention.

FIG. 6 is a diagram showing an example of a transfer function of a fourth digital filter of the active noise control system of the second embodiment.

FIG. 7 is a diagram showing an example of an impulse characteristic which is a transfer function of the FIR adaptive digital filter of the active noise control devices of the first and second embodiments.

FIG. 8 is a diagram showing an example of an impulse characteristic which is a transfer function of a fourth digital filter of the active noise control system of the second embodiment.

FIG. 9 is a diagram showing a result of a convolution calculation of the impulse characteristic of the FIR adaptive digital filter and the impulse characteristic of the fourth digital filter of the active noise control system of the second embodiment.

FIG. 10 is necessary for normal noise control within 1 sample,
It is the figure which showed the comparison with the time which concerns on the calculation of a digital filter, and the open time.

FIG. 11 is a system block diagram of a conventional active noise control device.

FIG. 12 is a system block diagram of a conventional active noise control device.

13 is a diagram showing the operation and filter characteristics of the system of the conventional example shown in FIG.

FIG. 14 is a diagram showing amplitude characteristics of a Butterworth filter.

FIG. 15 is a diagram showing a delay characteristic of a Butterworth filter.

FIG. 16 is a diagram showing amplitude characteristics of a Bessel type filter.

FIG. 17 is a diagram showing a delay characteristic of a Bessel type filter.

[Explanation of symbols]

 2 preamplifier 3 low-pass filter 4 A / D converter 5 first FIR digital filter 6 addition point 7 acoustic feedback suppression digital filter 8 FIR adaptive digital filter 9 second FIR digital filter 10 control algorithm 11 preamplifier 12 low-pass filter 13 A / D Converter 14 Third FIR Digital Filter 15 D / A Converter 16 Low Pass Filter 17 Power Amplifier 18 Fourth Digital Filter

─────────────────────────────────────────────────── ─── Continuation of the front page (51) Int.Cl. ⁶ Identification code Office reference number FI technical display location H04R 3/00 310

Claims (7)

[Claims] 1. An active noise in which a sound wave having a phase opposite to that of a sound wave propagating from a noise source in the sound wave propagation path and having the same sound pressure is generated, and the sound wave is muted at a predetermined position in the propagation path by the sound wave interference. A control device, wherein a state of interference between a first electromechanical converter that detects noise, an electromechanical converter that emits a sound wave to cancel at a predetermined position, and a sound wave emitted from the electromechanical converter is provided. The second electromechanical converter for detection and the analog signals from the first and second electromechanical converters are converted into digital signals, and the digital outputs are converted into analog signals and output to the electromechanical converters. An input / output interface, a FIR adaptive digital filter and an input signal that are output from the first electromechanical converter, and a second electromechanical filter. The coefficient of the FIR adaptive digital filter is digitally calculated based on the output signal of the gas converter and the calculation result of the first digital filter so that the output signal of the second mechanical-electrical converter becomes small.
In an active noise control device having a control algorithm for modifying the transfer function of an FIR adaptive digital filter,
An active noise control device characterized in that data obtained by arithmetic processing with a specific digital filter is used as input data for data obtained by converting analog signals from the first and second electromechanical converters into digital signals. . 2. An active noise in which a sound wave having a phase opposite to that of a sound wave propagated from a noise source in the sound wave propagation path and having the same sound pressure is generated, and the sound wave is muted at a predetermined position in the propagation path by the sound wave interference. A control device, wherein a state of interference between a first electromechanical converter that detects noise, an electromechanical converter that emits a sound wave to cancel at a predetermined position, and a sound wave emitted from the electromechanical converter is provided. The second electromechanical converter for detection and the analog signals from the first and second electromechanical converters are converted into digital signals, and the digital outputs are converted into analog signals and output to the electromechanical converters. An input / output interface, a FIR adaptive digital filter and an input signal that are output from the first electromechanical converter, and a second electromechanical filter. The coefficient of the FIR adaptive digital filter is digitally calculated based on the output signal of the gas converter and the calculation result of the first digital filter so that the output signal of the second mechanical-electrical converter becomes small.
In an active noise control device having a control algorithm for modifying the transfer function of an FIR adaptive digital filter,
An active noise control device characterized in that a transfer function of a specific digital filter is convoluted with a transfer function of an FIR adaptive digital filter modified by a control algorithm. 3. An active noise in which a sound wave having the same sound pressure as a phase opposite to that of a sound wave propagating from a noise source in the sound wave propagation path is generated, and the sound wave is silenced at a predetermined position in the propagation path by the sound wave interference. A control device, wherein a state of interference between a first electromechanical converter that detects noise, an electromechanical converter that emits a sound wave to cancel at a predetermined position, and a sound wave emitted from the electromechanical converter is provided. The second electromechanical converter for detection and the analog signals from the first and second electromechanical converters are converted into digital signals, and the digital outputs are converted into analog signals and output to the electromechanical converters. An input / output interface, a FIR adaptive digital filter and an input signal that are output from the first electromechanical converter, and a second electromechanical filter. The coefficient of the FIR adaptive digital filter is digitally calculated based on the output signal of the gas converter and the calculation result of the first digital filter so that the output signal of the second mechanical-electrical converter becomes small.
In an active noise control device having a control algorithm for modifying the transfer function of an FIR adaptive digital filter,
The delay time of the result obtained by convoluting the transfer function of the FIR adaptive digital filter modified by the control algorithm and the transfer function of the specific digital filter is calculated as the delay time of the transfer function of the FIR adaptive digital filter modified by the control algorithm. An active noise control device characterized in that the result of time shift or circulation so as to be time is used as a transfer function of a final FIR adaptive digital filter. 4. The active noise control device according to claim 2, wherein when the amount to be modified by the control algorithm becomes small, the convolution operation of the transfer function of the specific digital filter is started. 5. The convolution operation of the transfer function of a specific digital filter is started when the output signal of the second electromechanical converter reaches a certain level, and the convolution operation is started. The active noise control device described. 6. The active noise control device according to claim 2, wherein the transfer function of the FIR adaptive digital filter modified by the control algorithm is the transfer function of the specific digital filter. The active noise control device is characterized in that the time zone for performing the convolution calculation processing is a collection of time bodies other than the time body processed by the control algorithm. 7. The active noise control device according to claim 1, wherein the characteristic of the specific digital filter is a high-pass filter, a band-pass filter, or a low-pass filter. Active noise control device.