JPH0915035A - Adaptive signal processing method and equipment therefor, detection method for noise or vibration contribution, transfer characteristics-estimation method for noise or vibration, suppression method to noise or vibration, and simulation method for noise or vibration environments - Google Patents

Abstract

PURPOSE: To provide an adaptive signal processing method which can precisely analyze the impulse response of an acoustic system or a vibration system, and an equipment using the method. CONSTITUTION: The respective signals of noise or vibration which are detected at a plurality of points are inputted in first stage adaptive filters 431-433. The coefficients of the adaptive filters 431-433 are updated so that the difference e1 between the sum of the filter output signals and the noise signal detected at an estimation point where noise or vibration is estimated becomes minimum. Detection signals at a plurality of points are inputted in second stage adaptive filters 434-436, which are so updated that the difference e2 between the sum of the output signals and the difference e1 becomes minimum. The respective adaptive filter inputs are delayed through delay circuits 421-426. Impulse responses of various kinds of acoustic fields can be expressed by installing two stages of adaptive filters and combining the adaptive filters with delay time.

Description

Detailed Description of the Invention

[0001]

BACKGROUND OF THE INVENTION The present invention defines the characteristics of an adaptive digital filter so as to synthesize a signal at an evaluation point from one or a plurality of signals, and thereby analyzes the impulse response of an acoustic system or a vibration system. Adaptive signal processing method and device, and method for detecting noise or vibration contribution rate using the method, noise or vibration transfer characteristic estimation method, noise or vibration suppression method, and noise or vibration environment simulation method Is.

[0002]

2. Description of the Related Art Conventionally, estimation of spatial transfer characteristics of sound waves,
A number of methods using adaptive signal processing have been proposed for measuring and reducing noise and vibration from devices and the like. Among them, Japanese Patent Laid-Open No. 5-26722 is known as a conventional example of a contribution diagnosis method for investigating the degree of influence of a sound source or a vibration source. This is detected by inputting each noise or vibration signal detected at multiple points around the device, etc. to the adaptive filter, and the sum of the output signal of the adaptive filter and the evaluation point for evaluating noise or vibration. The coefficient of the adaptive filter is updated so that the residual difference with the noise or vibration signal is minimized, and the output signal of each adaptive filter when it converges to a constant value is used for noise or vibration at the evaluation point. The degree of contribution of the sound source or the vibration source is analyzed.

Further, in estimating the spatial transfer characteristic of a sound wave, a signal from a signal generation source is input to an adaptive filter, and an output signal of the adaptive filter and a signal from the signal generation source are radiated into space as a sound wave, The coefficient of the adaptive filter is updated so that the residual signal of the difference from the noise information signal detected at the installed measurement point is minimized, the filter coefficient is converged, and measurement is performed using the converged adaptive filter coefficient. The transfer characteristics of the space to the point were estimated.

[0004]

In the conventional adaptive signal processing method described above, the impulse response cannot be accurately estimated in an environment with a long reverberation time such as a high reverberation field. However, there is a problem that the residual difference from the noise or vibration signal detected at the noise or vibration evaluation point is not fully applied and a large amount remains. Furthermore, when the time interval between the direct wave and the reflected wave is long, it is necessary to increase the number of taps of the adaptive filter to represent the reflected wave.

It is an object of the present invention to provide an adaptive signal processing method and apparatus and a method therefor, which can accurately measure noise and vibration in a highly reverberant field and diagnose contribution of sound source and vibration source. Noise or vibration contribution detection method used, noise or vibration transfer characteristic estimation method, noise or vibration suppression method,
And to provide a method of simulating a noise or vibration environment.

[0006]

According to the present invention, a first one or a plurality of time series signals is input to one or a plurality of first stage adaptive filters, and the output signals of the adaptive filters are output. The coefficient of the first-stage adaptive filter is updated so that the first residual signal representing the difference between the sum and the second time-series signal different from the first time-series signal is minimized, The first time series signal is input to each of one or a plurality of second stage adaptive filters, and the sum of the output signals of the adaptive filters and the first
Disclosed is an adaptive signal processing method, characterized in that the second-stage adaptive filter is updated so that the second residual signal representing the difference from the residual signal is updated.

Further, the present invention is characterized in that the first time-series signal input to the first-stage and second-stage adaptive filters can be delayed by delay means having a variable delay time. A method for adaptive signal processing is disclosed.

According to the present invention, a plurality of noises or vibrations obtained by detecting the first time-series signal at a plurality of measurement points provided around a noise or vibration source using the above adaptive signal processing method. And the second time-series signal is a noise or vibration detection signal detected at an evaluation point provided for evaluating noise or vibration from the noise or vibration source. Signal processing is performed, and from each output of the adaptive filter at the end of the processing, a contribution rate of a detection signal at the measurement point corresponding to the output to a detection signal at the evaluation point is obtained. A method of detecting a contribution rate of noise or vibration is disclosed.

According to the present invention, by using the above adaptive signal processing method, the first time series signal is a noise or vibration signal to be applied to a noise or vibration generating means, and the second time series signal is , Adaptive signal processing is performed as a detection signal detected at an evaluation point for evaluating noise or vibration generated from the noise or vibration generating means, and the noise is calculated from the coefficient of each adaptive digital filter at the end of the processing. Alternatively, a noise or vibration transfer characteristic estimation method is disclosed, in which the noise or vibration transfer characteristic reaching the evaluation point is estimated from the vibration generating means.

Further, according to the present invention, by generating noise or vibration from a noise or vibration generating means for compensation, noise or vibration for suppressing noise or vibration from a noise or vibration source at an evaluation point. In the suppression method, the transfer characteristic of noise or vibration reaching the evaluation point is estimated from the noise or vibration generating means for compensation by using the above-mentioned method for estimating transfer characteristic of noise or vibration, and the transfer characteristic of noise or vibration is generated. A signal obtained by processing the reference noise or vibration detected in the vicinity of the source with the estimated transfer characteristic and the signal detected at the evaluation point are used for an adaptive filter having the reference noise or vibration as an input. A noise or vibration suppressing method characterized in that the noise or vibration at the evaluation point is suppressed by adjusting a coefficient and using the output of the adaptive filter as an input to the compensating noise or vibration generating means. To disclose.

Further, according to the present invention, the noise or vibration environment at the evaluation point is simulated by generating the noise or vibration at the evaluation point from the detection signals detected at a plurality of measuring points provided around the noise or vibration source. In a method for simulating a noise or vibration environment for generating, detecting the contribution rate at the evaluation point of the detection signal detected from each of the measurement points using the noise or vibration contribution rate detection means described above, Noise that generates noise or vibration toward the evaluation point using the detected contribution rate and the coefficient of the adaptive filter used when calculating the contribution rate, or a drive signal of a vibration generation means is generated. Alternatively, a method of simulating a vibration environment is disclosed.

[0012]

The adaptive filter is provided in two stages, and the delay means adjusts the delay time by providing a time difference between the adaptive filter in the first stage and the adaptive filter in the second stage. The target system can be approximated more accurately by the characteristics of the connected system, and in the case of a system in which the distance between the direct wave and the reflected wave is considerably long, the first-stage adaptive filter for the direct wave and the reflected wave for the reflected wave If a stage adaptive filter is used, the system can be approximated with a smaller number of taps.

By using the above method, the contribution rate of the measurement signal to the evaluation point and the transfer characteristic of the given noise or vibration transfer path can be detected or estimated more easily and accurately. Further, it is possible to effectively suppress the noise or vibration at the evaluation point by using the above estimated transfer characteristic,
Further, noise or vibration generated at the evaluation point can be simulated by using the detected contribution rate.

[0014]

Embodiments of the present invention will be described below with reference to the drawings. FIG. 2 is a diagram showing an overall configuration when the present invention is applied to a contribution rate analysis system. The contribution rate analysis system shown in FIG. 1 includes sensors 2a, 2b, 2c such as microphones for obtaining information on noise or vibration generated from the device 1.
And a microphone 3 installed at an evaluation point (near human ears) a certain distance away from the device 1, and sensors 2a, 2
Contribution rate analysis device 4 for analyzing how much the signals detected by b and 2c contribute to the microphone 3.
And a display unit 5.

FIG. 1 is a block diagram showing an embodiment of the contribution rate analyzing device 4 described above. The noise or vibration signals obtained by the sensors 2a, 2b and 2c and the microphone 3 are shown in FIG.
The noise signals obtained in
1 to 404 limits the band, and further A / D converter 41
At 1 to 414, it is converted into a digital signal. Sensor 2
The digital signals based on a, 2b, and 2c are 1
Stage adaptive digital filters (ADF) 431-43
3 and the output signals y1, y2 and y3 are generated by performing a convolution operation with the filter coefficient. At this time, the delay circuits 421 to 423 delay the digital signal as necessary. Next, the adaptive digital filter 4
The outputs y1 to y3 of 31 to 433 are added by the adding circuit 451 to be S1, and the difference signal e1 between this and the digital signal d obtained by digitizing the noise signal from the microphone installed at the evaluation point is given by the subtracting circuit 46. It is calculated. The adaptive algorithm circuits 441 to 443 set the adaptive digital filters 431 to 43 so that the difference signal e1 is minimized.
The coefficient of 3 will be updated.

Further, the digital signals obtained from the sensors 2a, 2b and 2c are delayed by delay circuits 424 to 426 for a predetermined time if necessary, and then the second-stage adaptive digital filters (ADF) 434 to. It is input to 436, and convolution calculation with the filter coefficient is performed here to generate output signals y4, y5, y6. Here, the delay time of the delay circuits 424 to 426 is the delay setting circuit 471.
Is set in association with the delay time of the first-stage delay circuits 421 to 423. Next, the adaptive digital filter 434
The outputs y4 to y6 of ˜436 are added by the adder circuit 452 to be s2, and the difference signal e1 which cannot be processed by the adaptive digital filters 431 to 433 of the first stage is added to s2.
The subtraction circuit 462 calculates the difference signal e2 between The adaptive algorithm circuits 444 to 446 adjust the adaptive digital filters 434 to 43 so that the difference signal e2 is minimized.
The coefficient of 6 is updated. The residual determination circuit 72 is
A threshold value for the magnitude of the difference signal e1 based on the first-stage adaptive digital filter is set, and if the difference signal e1 is larger than the threshold value, the second-stage adaptive digital filter is used, and if it is smaller, not used. It is something to do.

The above operation is repeated, and when the coefficients of the first-stage and second-stage adaptive digital filter systems converge to a constant value, from the outputs y1 to y6 of the respective adaptive digital filters 431 to 436, each sensor corresponding to the evaluation point. Analyze the contribution rate of noise or vibration from the installation position. The analyzed contribution rate, the coefficient of each adaptive digital filter, etc. are output to the display unit 5 and displayed.

As a method of analyzing the contribution rate, the sensor 2 is used.
The powers of the adaptive digital filter output signals corresponding to a, 2b, and 2c are set to the individual adaptive digital filters 431.
The value obtained by dividing the power of each of the output signals y1 to y6 of ˜436 and the difference signal e2 is the contribution rate for each sensor. For example, the contribution rate of the sensor 2a is expressed by the following equation.

(Equation 1)

According to this embodiment, the delay circuits 421 to 423 are provided before inputting to the first-stage adaptive digital filters 431 to 436, whereby the delay time can be adjusted. Therefore, in the case of a system in which it takes a long time for a direct wave to reach, the impulse response that can be represented by the same number of taps of the adaptive digital filter can be lengthened by delaying the delay circuit by the time delay. Also,
By providing the delay circuits 424 to 426 before inputting to the second-stage adaptive digital filters 434 to 436, in combination with the first-stage adaptive digital filters,
It can express impulse response in various sound environments. In addition,
The delay time of these delay circuits is set by varying the delay time of the adaptive digital filter to adjust the coefficient of the adaptive digital filter so that the error becomes smaller.

FIG. 3 shows an example of the impulse response. In a system where the time interval between the direct wave and the reflected wave is long as shown in this figure, the delay circuit should be used to determine the impulse response by dividing the direct wave from the first-stage adaptive filter and the reflected wave from the second-stage adaptive filter. It is possible to effectively use the number of taps. If the number of taps of the first-stage and second-stage adaptive digital filters can be variably set, the adaptive digital filters can be adjusted more accurately and easily in combination with the delay circuit. Furthermore, the number of adaptive filters is not limited to two, and a plurality of adaptive filters can be provided in parallel with three stages and four stages depending on the situation and surrounding environment. Further, the focusing state of the adaptive digital filter can be known by D / A converting the difference signals e1 and e2 and outputting as voice.

The present embodiment can be applied to general equipment that generates noise, such as a vacuum cleaner, an air conditioner, and a vehicle. Furthermore, the number of measurement points can be increased by increasing the number of sensors, speakers, and the like that obtain noise or vibration information. Further, it goes without saying that the present embodiment can be applied not only to the microphone and the vibration sensor but also to a sensor capable of detecting a time-series signal that fluctuates with time, such as an antenna or an optical sensor.

FIG. 4 is a block diagram showing an embodiment of the active muffler according to the present invention. This device includes a reference sensor 7 such as a microphone that obtains noise information from the device 6,
A speaker 8 which is an additional sound source for transmitting a sound wave (additional sound wave) having the same phase and an opposite phase as the sound wave propagating from the noise to a target position where noise is desired to be reduced, and a sound wave from the device 6 installed at the target position and a speaker Microphone 10 as a muffling error sensor for detecting the additional sound wave from 8 and controller 9 for controlling the generation of the additional sound wave from speaker 8.
Consists of

The controller 9 uses an estimated transfer function 93 between the input signal to the speaker 8 and the output signal of the microphone 10, which is necessary for the control here. FIG. 5 is a block diagram showing a measurement system for estimating the estimated transfer function 93, which is a system using the adaptive signal processing method of the present invention. And in FIG.
It is assumed that the speaker 8 and the microphone 10 are arranged at the same position in the same environment as that used in the actual sound reduction device of FIG. In FIG. 5, the reference signal s21 from the noise signal source 501 is converted into a digital reference signal s22 by the A / D converter 502, and the reference signal s22 is D /
Sound waves are emitted to the speaker 8 via the A converter 503 and the power amplifier 504 and emitted. The reference signal s22 is the first-stage adaptive digital filter (ADF).
The signal s23 is input to 505 and convolved with the filter coefficient to generate a signal s23. The sound wave radiated from the speaker 8 is detected by the microphone 10, and the A / D converter 5
It is digitized by 09 and becomes a digital signal s25. Difference signal e between the signal s25 and the signal s23
21 is obtained by the subtraction circuit 510, and the adaptive algorithm 507 updates the coefficient of the adaptive digital filter 505 so that the difference signal e21 becomes the minimum. At this time, the reference signal s22 is set by the delay circuit 511 as necessary.
Delay.

Further, the reference signal s22 is the delay circuit 5
The signal is delayed by a predetermined time by 12 and input to the second-stage adaptive digital filter (ADF) 506, and a convolution operation with the filter coefficient is performed to generate a signal s24. A subtraction circuit 513 obtains a difference e22 between the output signal s24 and the difference signal e21 that cannot be processed by the first-stage adaptive digital filter.
The adaptive algorithm 508 updates the coefficients of the adaptive digital filter 506 so that 22 is minimized. The above operation is repeated to converge the coefficients of each adaptive digital filter, and the estimated transfer function 93 is obtained from the converged values.

The operation of the active silencer of FIG. 4 using the estimated transfer function 93 obtained as described above will be described below. The detection signal s11 output from the reference sensor 7 as the noise information of the device 1 is digitized by the A / D converter 92 to be the reference signal s12, and the signal is obtained by performing the convolution operation of the adaptive digital filter 90. s1
Converted to 6. This signal s16 is the D / A converter 94.
Is converted into analog, is amplified by the power amplifier, and is emitted from the speaker 8. The microphone 10 detects the radiated sound and the sound wave to which the noise from the device 1 is added, and the output thereof is converted into a digital signal s14 by the A / D converter 95. Meanwhile, the reference signal s12 and the estimated transfer function 93
A virtual input signal s13 is generated by a convolution operation with, and the adaptive algorithm 91 updates the coefficient of the adaptive digital filter 90 using the signal s13 and the signal s14 so that the square value of the signal s14 is minimized. .

This embodiment can also be applied to general equipment that generates noise, such as air conditioners and vehicles. Further, in the above-described embodiment, the number of sensors, speakers, etc. for obtaining noise or vibration information can be increased. Furthermore, FIG.
The number of adaptive filters for obtaining the estimated transfer function is not only two, but a plurality of adaptive filters can be provided in parallel with three and four stages depending on the situation and the surrounding environment.

FIG. 6 is a block diagram showing an embodiment of a sound environment simulator using the adaptive signal processing method of the present invention. Sensors 12a, 12b such as microphones for obtaining information on noise or vibration of the device 11 as a noise source. , 12c, a microphone 13 installed at an evaluation point (near human ears) a certain distance away from the device 11, and sensors 12a, 12
b, 12c, a contribution rate analyzing device 14 for analyzing how much the signals detected by the b and 12c contribute to the microphone 13, and the contribution rates and the like obtained by the contribution rate analyzing device 14 are measured by the sensors 12a, 12b. The sound synthesizing means 15 for synthesizing the noise detected at 12c and synthesizing the noise at the evaluation point, and the speakers 16a, 1 for outputting the synthesized sound.
6b, and a dummy head 18 installed in a room 17 for reproducing a synthetic sound. Here, the configuration of the voice synthesis unit 15 is shown in FIG.

In this configuration, the contribution rate analyzing device 14
Is the same as that described in the embodiment of FIG.
The contribution rate of how much the noise detected by a to 12c contributes to the noise at the evaluation point detected by the microphone 13 is calculated and output. The contribution rate distribution circuit 151 of the voice synthesizing unit 15 determines the contribution rate of each of the sensors 12a-1.
The digital filter 15 is distributed to the contribution rate corresponding to 2c.
2a to 152c, respectively. Further, each adaptive digital filter coefficient at the time of convergence of the contribution rate analyzing means 14 is copied as a coefficient to the digital filters 152a to 152c. Each digital filter 152a-
In 152c, the convolution operation of the transfer function of the filter and the input contribution rate is performed, and the digital signals sa, s
b and sc are output. These signals are added to the addition circuit 153.
Are added and the combined signal s is input to the space correction circuit 154.

The storage medium 155 stores a database of measured values of transfer characteristics in a space having various transfer characteristics (reverberation characteristics, etc.), and the space correction circuit 154 stores:
The convolution operation of the transfer characteristic called from the storage medium 155 and the composite signal s is performed. Further, a convolution operation of the inverse filter 156 representing the inverse transfer characteristic from the speaker 16a, 16b in the room 17 for reproducing the synthetic sound to the human or the dummy head 18 to be evaluated and the output of the spatial correction circuit 154 is performed, and the operation result Is converted into an analog signal by the D / A converter and output from the speakers 16a and 16b.
In this way, product sounds in a space having various characteristics can be reproduced.

This embodiment can be applied to general equipment that generates noise, such as a vacuum cleaner, an air conditioner, and a vehicle. further,
It is possible to increase the number of sensors, speakers, and the like for obtaining information on noise or vibration. Furthermore, the number of adaptive filters in the contribution rate analyzing device 14 is not limited to two, and a plurality of adaptive filters may be provided in parallel with three or four stages depending on the situation and the surrounding environment.

[0031]

According to the present invention, in adaptive signal processing, two stages of adaptive filters are provided, and by combining each adaptive filter and delay time, impulse response of an acoustic system under various environments such as a high reverberant field. Can be calculated and the transfer function of the sound field can be estimated with high accuracy, and a muffling device and a sound environment simulator can be realized under the same environment.

[Brief description of the drawings]

FIG. 1 is a block diagram showing an embodiment of a contribution rate analyzing apparatus.

FIG. 2 is a block diagram showing an overall configuration of a contribution rate analysis system.

FIG. 3 is a diagram showing an example of an impulse response.

FIG. 4 is a block diagram showing an overall configuration of an active silencer.

5 is a diagram showing an embodiment of a system for obtaining an estimated transfer function in FIG.

FIG. 6 is an overall configuration diagram showing an embodiment of a sound environment simulator of the present invention.

FIG. 7 is a block diagram showing details of a voice synthesizing unit in FIG.

[Explanation of symbols]

 1, 6, 11 Devices that generate noise 2a, 2b, 2c, 7 Sensors 3, 10, 13 Microphones 4, 14 Contribution rate analyzers 421-426 Delay circuits 431-433 First-stage adaptive digital filters 434-436 2 Stage adaptive digital filter 451, 452 Adder circuit 461, 462 Subtractor circuit 481 Residual difference determination circuit 505, 506 Adaptive digital filter 510, 513 Subtractor circuit 511, 512 Delay circuit 8 Silence speaker 9 Controller 90 Adaptive digital filter 93 Estimated transfer Functions 16a, 16b Sound reproduction speaker 15 Sound synthesis means 18 Human or dummy head

 ─────────────────────────────────────────────────── ─── Continuation of the front page (72) Inventor Katsuo Oki 502 Jinritsu-cho, Tsuchiura-shi, Ibaraki Machinery Research Institute, Hiritsu Manufacturing Co., Ltd. Machinery Research Laboratory

Claims (14)

[Claims] 1. A first time series signal or a plurality of time series signals
The first or second plurality of first-stage adaptive filters are input to each of the first-stage adaptive filters, and a first time-sequential signal different from the sum of the output signals of the adaptive filter and the second time-series signal is displayed. The coefficient of the first-stage adaptive filter is updated so that the residual signal is minimized, and the first time-series signal is input to one or more second-stage adaptive filters, respectively, An adaptive signal processing method, wherein the second-stage adaptive filter is updated so that a second residual signal representing a difference between the sum of output signals of the filters and the first residual signal is minimized. 2. An adaptive filter provided in a third stage or an optional subsequent stage, which receives the first time-series signal as an input, and the coefficient of each adaptive filter is the sum of its output and the adaptive filter in the previous stage. The adaptive signal processing method according to claim 1, further comprising updating so as to minimize a difference from the residual signal generated by. 3. The first time series signal input to the first-stage and second-stage adaptive filters can be delayed by delay means having a variable delay time. 1. The adaptive signal processing method described in 1. 4. The adaptive signal processing method according to claim 3, wherein the delay time on the second-stage adaptive filter side is set in association with the delay time on the first-stage adaptive filter side. . 5. The delay time on the side of the first-stage adaptive filter is the first time obtained when the delay time is changed.
Is set to a value that minimizes the residual signal, and the delay time on the second-stage adaptive filter side is set to a value that minimizes the second residual signal obtained when the delay time is changed. The adaptive signal processing method according to claim 3, wherein the adaptive signal processing method is set. 6. The adaptive signal processing method according to claim 1, wherein each of the first-stage adaptive filter and the second-stage adaptive filter has a structure in which the number of taps is variable. 7. A threshold value for the first residual signal is provided, and the coefficient updating process in the second-stage adaptive filter is performed only when the residual signal is larger than the threshold value. The adaptive signal processing method according to claim 1, wherein: 8. A plurality of measurement points at which the first time-series signal is provided around a noise or vibration source by using the adaptive signal processing method according to any one of claims 1 to 7. Noise detected at a plurality of noises or vibrations, and the second time-series signal detected at an evaluation point provided for evaluating noises or vibrations from the noise or vibration source. Alternatively, adaptive signal processing is performed when a detection signal of vibration is detected, and the detection signal at the measurement point corresponding to the output is detected from each output of the adaptive filter at the end of the processing at the evaluation point. A method for detecting a contribution rate of noise or vibration, characterized by obtaining a contribution rate to a signal. 9. The noise or vibration contribution rate according to claim 8, wherein the first residual signal and the second residual signal are respectively D / A converted so that they can be auditioned as sounds. Detection method. 10. The adaptive signal processing method according to claim 1, wherein the first time-series signal is a noise or vibration signal to be applied to a noise or vibration generating means. And, the second time-series signal is subjected to adaptive signal processing as a detection signal detected at an evaluation point for evaluating noise or vibration generated from the noise or vibration generating means, and at the end of the processing A noise or vibration transfer characteristic estimating method for estimating a transfer characteristic of noise or vibration reaching the evaluation point from the noise or vibration generating means from a coefficient of each adaptive digital filter. 11. A noise or vibration suppressing method for suppressing noise or vibration from a noise or vibration source at an evaluation point by generating noise or vibration from a noise or vibration generating means for compensation. In the noise or vibration transfer characteristic estimating method according to claim 10, the noise or vibration transfer characteristic reaching the evaluation point is estimated from the compensating noise or vibration generating means to generate the noise or vibration. A signal obtained by processing the reference noise or vibration detected in the vicinity of the source with the estimated transfer characteristic and the signal detected at the evaluation point are used for an adaptive filter having the reference noise or vibration as an input. A noise or vibration suppressing method characterized by suppressing a noise or a vibration at the evaluation point by adjusting a coefficient and using an output of the adaptive filter as an input to the compensating noise or vibration generating means. . 12. A noise or vibration environment is generated by simulating noise or vibration at an evaluation point from detection signals detected at a plurality of measurement points provided around a noise or vibration source. In the method for simulating a noise or vibration environment, the noise or vibration contribution rate detecting means according to claim 8 is used to detect a contribution rate of the detection signal detected from each of the measurement points at the evaluation point, Noise that generates noise or vibration toward the evaluation point using the detected contribution rate and the coefficient of the adaptive filter used when calculating the contribution rate, or a drive signal of a vibration generation means is generated. Or a method of simulating a vibration environment. 13. One or a plurality of first stage adaptive filters for processing a first one or a plurality of time series signals, a sum of output signals of the adaptive filters and the first time series. First addition / subtraction means for calculating a difference from a second time-series signal different from the signal as a first residual signal; and the first stage adaptation so that the first residual is minimized. A first stage adaptive processing means for updating the coefficient of the filter; one or a plurality of second stage adaptive filters for processing the first time series signal; and a sum of output signals of the adaptive filter. Second adding / subtracting means for calculating a difference from the first residual signal as a second residual signal, and coefficients of the second-stage adaptive filter so that the second residual is minimized. An adaptive signal processing device comprising: a second stage adaptive processing means for updating; 14. A variable delay means for delaying the first time-series signal input to the first-stage and second-stage adaptive filters.
An adaptive signal processing device as described.