JPH07114391A - Active type noise controlling device and active type vibration controlling device - Google Patents

Abstract

PURPOSE:To eliminate a higher harmonic distortion without making a constitution complex by supplying a driving signal corrected based on a theoretically derived formula to a controlled sound source or a controlled vibration source. CONSTITUTION:A driving signal correcting part 12 is constituted of an arithmetic part 12a for calculating a value obtd. by cubing a driving signal ym generated by an adaptive digital filter Wm, an arithmetic part 12b for multiplying the resultant output by a correction coefficient a3 and outputting the result and an adding part 12c for adding the output of the arithmetic part 12b to the output of the adaptive digital filter Wm and taking the result as a corrected driving signal ym. That is, a driving signal is corrected by the driving signal correcting part 12 based on a formula y'=y+a3Xy3. Further, it is provided with a correction coefficient updating part 16 for updating the correction coefficient a3 of the arithmetic part 12b. The output of the adaptive digital filter Wm is corrected by the driving signal correction part 12, supplied to a loud speaker and the correction coefficient a3 within the driving signal correction part 12 is adaptively updated.

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 a vehicle for reducing noise by interfering control noise with noise transmitted from a noise source such as a vehicle engine into a space such as a passenger compartment. The present invention relates to an active vibration control device that reduces vibration by interfering control vibration with vibration that is emitted from a vibration source such as an engine and propagates through a vehicle body, and particularly relates to a loudspeaker or a control vibration as a control sound source with a simple configuration. The harmonic distortion can be removed from the output of an actuator or the like as a source.

[0002]

2. Description of the Related Art As conventional techniques of this kind, there are those described in British Patent No. 2149614 and Japanese Patent Publication No. 1-501344. These conventional devices are active noise reduction devices applied to aircraft cabins and similar closed spaces, in which a single noise source such as an engine located outside the closed space has a fundamental frequency f ₀ and It operates under the condition that noise including the harmonics f _{1 to} f _n is generated.

[0003] More specifically, includes a microphone for detecting a plurality of the installed sound pressure to a location within the closed space, and a plurality of loudspeakers for generating a control sound to the closed space, the frequency f ₀ of the noise source - Based on the f _n component, those frequencies f ₀ ~
The loudspeaker is driven by a signal having a phase opposite to that of the f _n component,
Therefore, a control sound having a phase opposite to that of the noise transmitted to the closed space is generated from the loudspeaker to cancel the noise.

Then, as a method of generating the control sound emitted from the loudspeaker, PROCEEDINGS OF THE IEEE, VOL.
63 PAGE 1692,1975, “ADAPTIVE NOISE CANCELLATION:
PRINCIPLES AND APPLICATIONS ”
An algorithm that applies the DROW LMS 'algorithm to multiple channels is applied. The content of the paper is "A MULTIPLE ERROR LMS ALGORITHM AND
ITS APPLICATION TOTHE ACTIVE CONTROL OF SOUND AND
VIBRATION ”, IEEE TRANS.ACOUST., SPEECH, SIGNAL PRO
CESSING, VOL.ASSP −35, PP. 1423−1434, 1987.

That is, the LMS algorithm is one of the algorithms suitable for updating the filter coefficient of the adaptive digital filter, and is, for example, a so-called Filter.
In the ed-X LMS algorithm, a transfer function filter that models the transfer function from the loudspeaker to the microphone is set for all combinations of the loudspeaker and the microphone, and a reference signal representing the noise generation state of the noise source is set. Update the filter coefficient of the adaptive digital filter with variable filter coefficient provided for each loudspeaker so that the value of the predetermined evaluation function based on the value processed by the filter and the residual noise detected by each microphone is reduced. is doing.

[0006]

Considering a case where the above noise reduction device is applied to a vehicle, for example,
First, it must be taken into consideration that the vehicle interior noise generally has a relatively large sound pressure even in a low frequency band below 100 Hz. For example, the engine rotation secondary component (muffled sound), which is the main component of the noise emitted from the 4-cycle 4-cylinder engine, becomes 100 Hz or less at 3000 rpm or less, and the vibration input from the wheels propagates through the vehicle body.
The noise also has the largest sound pressure in the vicinity of 30 to 40 Hz, which is called the drumming region. That is, it is necessary to mount a loudspeaker that does not deteriorate output characteristics even on the low frequency side.

On the other hand, a general cone type loudspeaker vibrates the air, which is a medium, to generate a sound pressure when the cone moves back and forth, so that the same sound pressure is generated in an ideal free sound field. Requires a constant volume velocity regardless of frequency, but a low frequency means that the cone moves back and forth less frequently per unit time, so the distance (stroke) that the cone moves back and forth once If is not increased, the same volume velocity as at the higher frequency cannot be generated. It should be noted that a closed space such as a vehicle compartment is not in exactly the same situation as the free sound field because of the influence of the acoustic mode and the like, but the tendency to require a larger stroke at low frequencies is the same.

However, in a general cone type speaker, as shown in FIG. 10, a frame 51 has an edge 52 acting as a spring element and a spider 5 acting as a damper element.
Since the cone 54 is fixed via 3 and 3, the supporting rigidity of the edge 52 and the spider 53 rapidly increases as they approach the limit of physical movement, and the air spring of the enclosure that covers the back side of the cone 54 increases. Since the rigidity of the cone 54 rapidly increases in a region where the stroke of the cone 54 is large, the cone 54 moves non-linearly in a range where the stroke amount is large, and the sound emitted from the speaker is distorted (non-linearity of the vibration system). It causes harmonic distortion).

Further, the cone 54 positions the voice coil 55 fixed thereto in the magnetic field generated by the magnet 58 sandwiched between the pole piece 56 and the plate 57, and controls the current flowing through the voice coil 55 appropriately. Therefore, there is no particular problem in a situation where the voice coil 55 exists in a region where the magnetic flux is uniform as shown in FIG. 11, but when the stroke of the cone 54 increases, the voice The coil 55 reaches a region where the magnetic flux distribution is non-uniform, and the cone 54 moves non-linearly, which also distorts the sound emitted from the speaker (harmonic distortion due to the nonlinearity of the drive system).
Will occur.

That is, due to the structure of the loudspeaker, FIG.
If the distortion is such that the upper and lower peaks of the output waveform are evenly dull with respect to the input waveform as shown in Fig. 5, odd-order harmonics (3 times, 5 times, 7 times the input signal frequency, ...) Appear. If the distortion is such that one of the upper and lower peaks is blunted as shown in FIG. 13 due to superposition of DC components, even harmonics (2 times, 4 times, 6 times the input signal frequency, ... There is a problem that) appears.

The problem that the output is distorted is not limited to the loudspeaker.
An active vibration reduction device that generates a control vibration by displacing a movable plate by an electromagnetic solenoid or the like (see, for example, Japanese Patent Laid-Open No. Hei 2
-42228 publication. ) Occurs for the same reason. The above-mentioned problems are caused by the cone 54
Can be solved by increasing the range of linear movement, specifically by using a large loudspeaker, but there is a limit because it is a structural defect, and an extremely large loudspeaker is a vehicle If there is a small space as in the above, it may not be possible to mount the device. As a measure for solving such a problem, there is a motional feedback speaker (MFB speaker) (for example, "SANYO
TECHNICAL REVIEW VOL.13 NO.2 AUG.1981 "p37-4
Detailed in 5. ), Since the MFB speaker requires the addition of a vibration pickup and a feedback circuit, it may cause an increase in cost and a decrease in reliability, and particularly when the amplifier and the speaker are arranged apart from each other like a vehicle. In order to feed back a signal, a new harness is required, resulting in a significant increase in cost and a decrease in reliability.

The present invention has been made paying attention to the unsolved problems of the prior art as described above, and the output as described above can be achieved without causing a significant increase in cost or a decrease in reliability. An object of the present invention is to provide an active noise control device and an active vibration control device capable of removing waveform distortion.

[0013]

In order to achieve the above object, the invention according to claim 1 is a basic configuration diagram thereof.
As shown in, a control sound source capable of generating a control sound in a space where noise is transmitted from the noise source, and a reference signal generation unit that detects a noise generation state of the noise source and outputs the reference signal,
Residual noise detecting means for detecting residual noise at a predetermined position in the space and outputting it as a residual noise signal, and an adaptive digital filter with a variable filter coefficient for filtering the reference signal to generate a drive signal for driving the control sound source. And an adaptive processing means for updating the filter coefficient of the adaptive digital filter so as to reduce noise in the space based on the reference signal and the residual noise signal, the active noise control device comprising: Drive signal correction means for correcting the drive signal before being supplied to the sound source based on the following equation (1) is provided.

According to a second aspect of the present invention, in the active noise control device according to the first aspect of the present invention, correction coefficient updating means for updating the correction coefficient a ₃ based on the following equation (4) is provided. It was In order to achieve the above purpose,
In the invention according to claim 3, as shown in FIG. 2 which is a basic configuration diagram thereof, a control sound source capable of generating a control sound in a space where noise is transmitted from the noise source and a noise generation state of the noise source are detected. Reference signal generating means for outputting as a reference signal, residual noise detecting means for detecting residual noise at a predetermined position in the space and outputting as a residual noise signal, and driving the control sound source by filtering the reference signal. An adaptive digital filter having a variable filter coefficient for generating a drive signal,
An active noise control device comprising: an adaptive processing unit that updates the filter coefficient of the adaptive digital filter so that noise in the space is reduced based on the reference signal and the residual noise signal. Drive signal correction means for correcting the drive signal before being supplied based on the following equation (2) is provided.

According to a fourth aspect of the present invention, in the active noise control device according to the third aspect of the invention, the correction coefficients a ₂ and a ₃ are updated based on the following equations (3) and (4). The correction coefficient updating means is provided. On the other hand, in order to achieve the above-mentioned object, the invention according to claim 5 is a control vibration source capable of generating a control vibration that interferes with a vibration emitted from a vibration source, and a vibration generation state of the vibration source is detected and a reference A reference signal generating means for outputting as a signal, a residual vibration detecting means for detecting the residual vibration after the interference and outputting as a residual vibration signal, and a drive signal for filtering the reference signal to drive the control vibration source. An adaptive digital filter having a variable filter coefficient to be generated, and an adaptive processing means for updating the filter coefficient of the adaptive digital filter based on the reference signal and the residual vibration signal so as to reduce the vibration after the interference are provided. The active vibration control device is provided with drive signal correction means for correcting the drive signal before being supplied to the controlled vibration source based on the following equation (1). That.

According to a sixth aspect of the present invention, in the active vibration control device according to the fifth aspect of the invention, correction coefficient updating means for updating the correction coefficient a ₃ based on the following (4) is provided. The active vibration control device according to claim 5. In order to achieve the above object, the invention according to claim 7 is a control vibration source capable of generating a control vibration that interferes with a vibration emitted from a vibration source, and a vibration generation state of the vibration source is detected to be a reference. A reference signal generating means for outputting as a signal, a residual vibration detecting means for detecting the residual vibration after the interference and outputting as a residual vibration signal, and a drive signal for filtering the reference signal to drive the control vibration source. An adaptive digital filter having a variable filter coefficient to be generated, and an adaptive processing means for updating the filter coefficient of the adaptive digital filter based on the reference signal and the residual vibration signal so as to reduce the vibration after the interference are provided. The active vibration control device is provided with drive signal correction means for correcting the drive signal before being supplied to the controlled vibration source based on the following equation (2). That.

The invention according to claim 8 is the active vibration control device according to claim 7, wherein the correction coefficients a ₂ and a ₃ are updated based on the following equations (3) and (4). The correction coefficient updating means is provided. _{y '= y + a 3 ×} y 3 ...... (1) y' = y + a 2 × y 2 + a 3 × y 3 ...... (2) a 2 (n + 1) = ν 2 × a 2 (n) -μ 2 × e (N) × t ₂ (3) a ₃ (n + 1) = ν ₃ × a ₃ (n) -μ ₃ × e (n) × t ₃ (4) However, after y'is corrected Drive signal, y is a drive signal before being corrected, a ₂ and a ₃ are correction coefficients, ν ₂ and ν ₃ are divergence suppressing coefficients, μ ₂ and μ ₃ are convergence coefficients, and e is a residual noise signal or residual vibration. Signal, (n), (n + 1)
The terms with a mark represent values at discrete times n and n + 1, respectively.

Further, t ₂ and t ₃ are values obtained by the convolution operation defined by the following equations (5) and (6), and C ^ _i is between the control sound source and the residual noise detecting means or the control vibration. The transfer function C between the source and the residual vibration detecting means is modeled in the form of a finite impulse response function. The i-th filter coefficient of the transfer function filter C ^, J is the number of taps of the transfer function filter C ^ (the number of filter coefficients). Is.

[0019]

[0020]

First, the odd-order distortion as shown in FIG.
If the terms higher than the third order are ignored, it can be approximated by the following equation (7). v (n) = y (n) + b ₃ × y (n) ³ (7) where y (n) is the drive signal at the discrete time n, v
(N) is a signal corresponding to the actual movement (acceleration) of the speaker cone, b ₃ is a third-order distortion coefficient, and b ₃ <0 because it acts in a direction that makes the cone difficult to move.

Next, the drive signal y in equation (7)
A signal y ′ (n) obtained by previously processing (n) with the following equation (8).
Consider the case of replacing with. y ′ (n) = y (n) + a ₃ × y (n) ³ (8) That is, the drive signal y (n), which is the output of the adaptive digital filter, is not directly supplied to the control sound source,
The value corrected based on the cube of the drive signal y (n) as in the above equation (8) is considered as a new drive signal y ′ (n).

Then, the signal v (n) representing the actual movement of the speaker cone is expressed as follows. v (n) = y ′ (n) + b ₃ × y ′ (n) ³ = y (n) + a ₃ × y (n) ³ + b ₃ (y (n) + a ₃ × y (n) ³ ) ³ = y (n) + (a ₃ + b ₃ ) y (n) ³ + 3a ₃ × b ₃ × y (n) ⁵ + 3a ₃ ² × b ₃ × y (n) ⁷ + a ₃ ³ × b ₃ × y (n) ⁹ (10) Then, in this equation (10), the higher-order terms higher than the fifth order are extremely smaller than the terms lower than the third order, so if they are ignored, the drive signal y (n) and the signal v (n It can be seen that the difference with) is minimum when a ₃ = −b ₃ (11).

From the above, in the invention according to claim 1,
The drive signal correction means uses the drive signal y based on the equation (1).
Is corrected, the odd-order harmonic distortion included in the control sound emitted from the control sound source is removed by appropriately selecting the correction coefficient a ₃ . Furthermore, (11) above
Consider that the correction coefficient a ₃ that satisfies the expression is adaptively obtained. That is, since the drive signal y (n) is a signal obtained by filtering the reference signal x (n) generated by the reference signal generating means with the adaptive digital filter W, it is expressed by the following equation (12). Represented by.

[0024] However, W _i is the i-th filter coefficient of the adaptive digital filter W, and I is the number of taps of the adaptive digital filter W. The sound emitted from the control sound source and reaching the residual noise detecting means is represented by the following equation (13).

[0025] The residual noise signal e (n) is the sound d (n) arriving from other than the noise source and the sound o (n) emitted from the control sound source.
Since and are superposed, they are expressed as the following equation (14).

E (n) = d (n) + o (n) (14) Here, the evaluation function J is defined as J = e (n) ² (15), and this evaluation function J is the minimum. Think about
That is, when the evaluation function J is differentiated by the correction coefficient a ₃ , Therefore, ignoring the higher-order terms of order 5 or higher in this equation (16), Becomes

Therefore, the update formula of the correction coefficient a ₃ according to the LMS algorithm is: a ₃ (n + 1) = ν ₃ a ₃ (n) -μe, where divergence suppression coefficient is ν ₃ and convergence coefficient is μ _3. (N) t ₃ , which is equal to the equation (4). It should be noted that the divergence suppression coefficient is a coefficient for preventing the updated value from diverging due to an unexpected cause (for example, the effect of ignoring higher-order terms), and is usually slightly more than 1. Although a small value is adopted, ν ₃ = 1 may be set in the case of a stable system with a small possibility of divergence. The convergence coefficient is a coefficient that contributes to the speed of convergence and the stability at the time of convergence.

From the above, in the invention according to claim 2, since the correction coefficient a ₃ adaptively converges to the optimum value, the odd harmonic distortion included in the control sound emitted from the control sound source is surely generated. Will be removed. Here, since an ordinary speaker cone is designed to have the same characteristics in both the forward and backward directions, there may be a case where odd-order harmonic distortion in which the upper and lower peaks of the output waveform are evenly dull as shown in FIG. In most cases, even-order harmonic distortion as shown in FIG. 13 may occur if the characteristics in the forward and backward directions are different, or if the DC components are superposed even if they are equal.

Therefore, consider a case where the drive signal y (n) is replaced with a signal y '(n) processed in advance by the following equation (18). y ′ (n) = y (n) + a ₂ y (n) ² + a ₃ y (n) ³ (18) Then, the signal v (n) is: v (n) = y ′ (n) + b ₂ y '(n) ² + b ₃ y' (n) ³ = y (n) + a ₂ y (n) ² + a ₃ y (n) ³ + b ₂ {y (n) ² + a ₂ ² y (n) ⁴ + a ₃ ² y (n) ⁶ + 2a ₂ y (n) ³ + 2a ₂ a ₃ y (n) ⁵ + 2a ₃ y (n) ⁴ } + b ₃ {y (n) ³ + a ₂ ² y (n) ⁵ + a ₃ ² y (n) ⁷ + 2a ₂ y (n) ⁴ + 2a ₂ a ₃ y (n) ⁶ + 2a ₃ y (n) ⁵ + a ₂ y (n) ⁴ + a ₂ ³ y (n) ⁶ + a ₂ a ₃ ² y ( n) ⁸ + 2a ₂ ² y (n) ⁵ + 2a ₂ ² a ₃ y (n) ⁷ + 2a ₂ a ₃ y (n) ⁶ + a ₃ y (n) ⁵ + a ₂ ² a ₃ y (n) ⁷ + a ₃ ^{3 _{y (n) 9 + 2a 2}} a 3 y (n) 6 + 2a 2 a 3 2 y (n) 8 + 2a 3 2 y (n) 7} = y (n) + (a 2 b ₂₎ it becomes ^{y (n) 2 + (a} 3 + 2a 2 b 2 + b 3) y (n) 3 + ... ...... (19).

From a ₂ <1, b ₂ <1, a ₂ b ₂
«1 and disregard the terms higher than the third order because they have a small effect, so that drive signal y (n)
The difference between the signal v (n) _{is, a 2 = -b 2 a 3} = It can be seen that a minimum when the -b _3.

From the above, in the invention according to claim 3,
The drive signal correction means uses the drive signal y based on the above equation (2).
Is corrected, the odd-order harmonic distortion and even-order harmonic distortion included in the control sound emitted from the control sound source are removed by appropriately selecting the correction coefficients a ₂ and a ₃ . Further, the update equations of the correction coefficients a ₂ and a ₃ are as follows: a ₂ (n) = ν ₂ a ₂ (n) −μ ₂ e (n) t ₂ a ₃ (n) = ν _{3 a 3 (n) -μ 3} e (n) t 3 , and the above (3), equal to (4) below. The specific contents of the divergence suppression coefficient ν ₂ and the convergence coefficient μ ₂ are the same as the divergence suppression coefficient ν ₃ and the convergence coefficient μ ₃ described above.

From the above, in the invention according to claim 4, since the correction coefficients a ₂ and a ₃ adaptively converge to the optimum values, the odd harmonics contained in the control sound emitted from the control sound source. Distortion Even-order harmonic distortion is reliably removed. Here, the inventions according to claims 1 to 4 are all directed to noise, whereas the inventions according to claims 5 to 8 are directed to vibration. Therefore, the operations of the inventions according to claims 5 to 8 are substantially the same as those of the inventions according to claims 1 to 4, although there is a difference between sound and vibration.

[0033]

Embodiments of the present invention will be described below with reference to the drawings. 3 and 4 are diagrams showing the configuration of the first embodiment of the present invention, in which the active noise control device according to the present invention is applied to a vehicle for reducing noise in a vehicle interior 3 This is applied to the active noise control device 1.

First, the structure will be described. As shown in FIG. 3, the vehicle active noise control system 1 operates as a noise transmitted from the engine 4 as a noise source into the vehicle interior 3 as a space. This is a device for reducing muffled sound, and a crank angle sensor 5 that outputs a crank angle signal CP that is a pulse signal synchronized with the rotation of the crank shaft of the engine 4 is attached to the engine 4.

On the other hand, a plurality of microphones 8a, 8b (only two of which are shown in FIG. 3) as residual noise detecting means for measuring the sound pressure of noise remaining in the vehicle compartment 3 are provided in the vehicle compartment 3.
A plurality of (only one is shown in FIG. 3) loudspeakers 9 as control sound sources for generating control sounds are arranged in the passenger compartment 3. The crank angle signal CP output from the crank angle sensor 5 and the residual noise signal e output from the microphones 8a and 8b are supplied to a controller 10 including a microcomputer or the like.

The controller 10 receives each signal C supplied.
Predetermined arithmetic processing is executed based on P and e, and the engine 4
The drive signal y is supplied to the loudspeaker 9 so that the loudspeaker 9 emits a control sound for canceling the muffled sound transmitted from the inside to the vehicle interior 3. As shown in FIG. 4, which is a block diagram showing the functional configuration of the controller 10, the controller 10 has a reference signal x which is a sinusoidal signal having the same cycle as the crank angle signal CP supplied from the crank angle sensor 5.
And the adaptive digital filter W _m (m = _m ) having a variable filter coefficient that generates the drive signal y _m by filtering the reference signal x (specifically, by performing convolution integration). 1, 2, ..., M:
M has the number of loudspeakers) and a drive signal correction unit 12 that corrects the drive signal y _m output from the adaptive digital filter W _m, and the result corrected by the drive signal correction unit 12 is the final result. Each loudspeaker 9 is supplied as a typical drive signal y _m .

Then, the drive signal correction unit 12 multiplies the drive signal y _m generated by the adaptive digital filter W _m by a cube, and the output of the calculation unit 12 a by the correction coefficient a ₃ . And the output of the adaptive digital filter W _m , and the output of the calculation unit 12 b is added to obtain the corrected drive signal y _m.
2c. That is, the drive signal correction unit 12 is configured to correct the drive signal y _m based on the equation (1). The drive signal correction unit 12
In reality, however, M drive signals are provided corresponding to M drive signals y _m , but one drive signal correction unit 12 is shown in FIG. 4 for the sake of clarity.

Further, the controller 10 uses the transfer function filter C ^ _{lm to} which the reference signal x is input, the value r _{lm obtained} by processing the reference signal x with the transfer function filter C ^ _lm , and the residual noise signal e based on the adaptive digital signal. The filter coefficient updating unit 15 that updates the filter coefficient W _mi of the filter W _m . The transfer function filter C _lm (l = 1, 2,
, L, m = 1, 2, ..., M) is a digital filter in which the transfer function C _lm between the loudspeaker 9 and the microphones 8a, 8b is modeled in the form of a finite impulse response function. Built for all combinations of L loudspeakers and L microphones (L × M).

[0039] Then, the filter coefficient updating unit 15, the residual noise signal e _l supplied from the processing signal r _lm and microphone 8a, 8b to be supplied from the transfer function filter C ^ _lm
According to, the filter coefficient W _mi of the adaptive digital filter W _m (i = 0, 1, 2, ..., I-1: I is the number of taps of the adaptive digital filter W _m ) It is designed to be updated based on equation (20).

[0040] Note that ν _w is a divergence suppression coefficient, μ _w is a convergence coefficient, and r _mi (n
-I) is a value defined by the following equation (21).

[0041] Furthermore, the controller 10 includes a drive signal correction unit 12
Of the transfer function filter C to which the output of the calculation unit 12a of
^ _Lm and the value t ₃ generated by its transfer function filter C ^ _lm
The correction coefficient updating unit 16 updates the correction coefficient a ₃ of the arithmetic unit 12b according to the above equation (4) based on (see the above equation (6)) and the residual noise signal e _l .

That is, in this embodiment, the output of the adaptive digital filter W _m is not directly supplied to the loudspeaker 9, but is corrected by the drive signal correction unit 12 before being supplied to the loudspeaker 9. In addition, the correction coefficient a ₃ in the drive signal correction unit 12 is adaptively updated. In the invention described in each claim, y '((n) is calculated in order to calculate the output signal y' (n) that reduces the third-order distortion without limiting the number of channels of the system. n) = y (n) + a ₃ y ³ (n) a ₃ (n + 1) = ν ₃ a ₃ (n) −μ ₃ e (n) t ₃ However, in the case of a system of multiple channels such as M loudspeakers and L microphones as in this embodiment, the calculation can be performed as follows.

[0043] _{y 'm (n) = y} m (n) + a 3m y 3 m (n) a 3m (n + 1) = ν 3 a 3m (n) -μ 3 Σe l (n) t
_3lm However, l is the microphone number, and l = 1, 2,
..., L and m are loudspeaker numbers, and m = 1, 2,
, M, C ^ _lmi is the i-th filter coefficient of the transfer function C ^ _lm from the m-th loudspeaker to the l-th microphone, e _l is the output of the l-th microphone, y ′
_m (n) is a control signal output to the m-th loudspeaker.

FIG. 5 is a flowchart showing the outline of the processing executed in the controller 10. The operation of this embodiment will be described below with reference to FIG. That is, the process shown in FIG. 5 is executed in synchronization with a predetermined sampling clock. First, in step 101, the reference at the current discrete time n is based on the input timing of the crank angle signal CP. Generate the signal x (n).

Then, the routine proceeds to step 102, where the reference signal x generated one after another each time the processing of FIG. 5 is executed.
(N) and each filter coefficient W _mi of the adaptive digital filter W _m are convoluted to generate the drive signal y _m . Then, the process proceeds to step 103, and according to the above equation (1),
The drive signal y _m is corrected. Specifically, a value y ³ that is the cube of the drive signal y _m is obtained, the result is multiplied by the correction coefficient a ₃ , the result is added to the original drive signal y _m, and the addition result is corrected. The generated drive signal y _m .

When the corrected drive signal y _m is obtained,
In step 104, the drive signal y _m is output to each loudspeaker 9. Next, the routine proceeds to step 105, where the residual noise signal e _l is read and at the same time, step 1
06, the reference signal x (n) and the transfer function filter C
The processing signal r _lm is calculated by _convolving each filter coefficient C ^ _{lmj of} ^ _lm , and the process _proceeds to step 107. Based on the residual noise signal e ₁ and the processing signal r _lm , according to the above equation (20). Each filter coefficient W _mi of the adaptive digital filter W _m is updated.

Next, in step 108, the value y ³ obtained when the drive signal y _m is corrected according to the above equation (6) and each filter coefficient C ^ of the transfer function filter C ^ _lm.
A value t ₃ is obtained by _{convolving lmj} with _lmj, and the _{process proceeds} to step 109, and the correction coefficient a ₃ is updated based on the value t ₃ and the residual noise signal e _l according to the above equation (4). When the processing of step 109 is completed, the processing of this time is terminated, and step 10 is performed at the timing of the next sampling clock.
The process is executed again from 1.

When such processing is executed, the drive signals y _m are supplied to the loudspeakers 9 one after another.
A control sound corresponding to the drive signal y _m is generated in the passenger compartment 3, but each filter coefficient W _mi of the adaptive digital filter W _m does not always converge to an optimum value immediately after the control is started. Therefore, it cannot be said that the muffled sound transmitted from the engine 4 into the vehicle interior 3 is reduced by the control sound emitted from the loudspeaker 9.

However, when the process shown in FIG. 5 is repeatedly executed, the filter coefficient updating section 15 causes the filter coefficient W of the adaptive digital filter W _m to be based on the LMS algorithm.
_{Since mi} is updated, the muffled noise is canceled by the control sound emitted from the loudspeaker 9, and the noise in the vehicle interior 3 is reduced. Moreover, in the present embodiment provides a filter output of the adaptive digital filter W _m as opposed to supplied to the loudspeaker 9 as a driving signal, a loudspeaker 9 as appropriate corrected by the drive signal correcting section 12 Since the correction content follows the theoretically derived equation (1), M
It is possible to remove the odd-order harmonic distortion included in the control sound emitted from the loudspeaker 9 without requiring a particularly complicated structure such as the FB speaker.

Then, the correction coefficient a ₃ for determining the correction amount is
Since the configuration is adaptively updated according to the theoretically derived equation (4), even if the characteristics of the loudspeaker 9 itself deteriorate, for example, the odd-order harmonic distortion included in the control sound can be ensured. Can be removed. Therefore, it is possible to improve the acoustic characteristics in the low frequency band without making the loudspeaker 9 particularly large, and to provide an active noise control device for a vehicle that is effective even for muffled sound of 100 Hz or less, for example. Can be done.

Here, in the present embodiment, the crank angle sensor 5, the reference signal generating section 11 and the processing of step 101 constitute the reference signal generating means, and the filter coefficient updating section 15 and the transfer function filter C provided in the preceding stage thereof. ^ _lm and adaptive processing means by the processing of step 105 to 107 is constituted by the processing of the drive signal correcting section 12 and the step 103 is configured driving signal correction means, the correction coefficient update unit 16, the transfer function provided in the preceding stage Filter C ^
_The correction coefficient updating means is configured by _lm and the processes of steps 108 and 109.

FIG. 6 is a conceptual diagram of a simulation performed by the inventors of the present invention. As shown in FIG. 6A, the amplitude of a sine-wave output waveform with an amplitude of 1 output from the oscillation circuit 20 is After being adjusted by the amplifier circuit 21 having a gain of either 1.0 or 1.2, it is supplied to the adaptive filter 22 whose filter coefficient is updated according to the LMS algorithm, and the output of the adaptive filter 22 is supplied to FIG. The output of the amplifier circuit 21 is directly supplied to the adder 25 while being supplied to the adder 25 via the speaker model 23 and the delay circuit 24 having the input / output characteristics in which an output with an amplitude of 1 or more as shown in FIG. The filter coefficient of the adaptive filter 22 is supplied so that the output of the adder 25 becomes smaller.

FIG. 7 is a diagram showing a simulation result when the gain of the amplifier circuit 21 is 1.0.
FIG. 7A shows the result when the output of the adaptive filter 22 is directly supplied to the speaker model 23 (conventional example).
(D) is the result when the same correction processing as in the present embodiment is performed (this embodiment). 7 (b) is an enlarged view of the range of the arrow in FIG. 7 (a), FIG. 7 (e) is an enlarged view of the range of the arrow in FIG. 7 (d), and FIG.
FIG. 4 is a frequency characteristic diagram showing the noise reduction effect according to the conventional example and the present example compared with the noise level before control.

That is, in the conventional example, as a result of being greatly influenced by the input / output characteristics of the speaker model 23, a frequency component three times as high as the output of the oscillation circuit 20 is large, but in this embodiment, the correction process is performed. Therefore, such triple frequency components can be reduced as compared with the conventional example, and a good noise reduction effect can be obtained for the fundamental frequency component.

FIG. 8 is a diagram showing a simulation result when the gain of the amplifier circuit 21 is 1.2.
8A and 8B show the results of the conventional example, and FIG.
(C) is the result of this example. However, FIG.
In (a), the divergence suppression coefficient ν _w and the convergence coefficient μ _w in the update calculation of the adaptive filter are ν _w = 1.0 and μ _w = 0.5, and in FIGS. 8 (b) and (c), ν _w = 0.99 and μ _w = 0.4.

That is, as can be seen from the comparison between FIGS. 8A and 8B, the divergence suppression coefficient ν is obtained even in the conventional example.
By setting _w appropriately, it is possible to reduce the noise level of the component of the frequency three times the fundamental frequency.
This has the drawback that the noise reduction effect at the fundamental frequency is not so great. On the other hand, with the configuration of this embodiment, the noise reduction effect of the fundamental frequency component can be made relatively large, while the noise level of the component having a frequency three times the fundamental frequency can be further reduced.

FIG. 9 is a diagram showing a second embodiment of the present invention, and is a block diagram showing a functional configuration of the controller 10 similarly to FIG. 4 of the first embodiment. Since the overall structure is the same as that of the first embodiment, its illustration and description will be omitted, and the same structures as those of the first embodiment will be denoted by the same reference numerals and their duplicate description will be omitted. That is, in the present embodiment, in addition to the configuration of the first embodiment, the drive signal correction unit 12 includes a calculation unit 12d that calculates a squared value of the drive signal y _m generated by the adaptive digital filter W _m . An arithmetic unit 12e that multiplies the output of the arithmetic unit 12d by the correction coefficient a ₂ and outputs the result is provided, and the addition unit 12c uses the pre-correction drive signal y _m , the arithmetic result of the arithmetic unit 12b, and the arithmetic unit 12e. And the result of addition are output, and the addition result is output as the corrected drive signal y _m .

Further, the controller 10 includes an arithmetic unit 12
The transfer function filter C ^ _lm to which the output of d is input and the value t ₂ generated by the transfer function filter C ^ _lm ((5) above).
The correction coefficient updating unit 17 updates the correction coefficient a ₂ of the calculation unit 12e based on the equation (3) and the residual noise signal e _l according to the above equation (3). That is, in the present embodiment, the drive signal correction unit 12 is configured to correct the drive signal y _m based on the theoretically derived equation (2), and therefore the drive signal y _m is emitted from the loudspeaker 9. Both the odd-order harmonic distortion and the even-order harmonic distortion included in the control sound can be removed.

Moreover, since the correction coefficients a ₂ and a ₃ which determine the correction amount are adaptively updated according to the theoretically derived equations (3) and (4), the odd number included in the control sound Both the second harmonic distortion and the even harmonic distortion can be reliably removed. Other functions and effects are similar to those of the first embodiment. here,
In this embodiment, the correction coefficient updating unit 16, the transfer function filter C ^ _lm provided in the preceding stage, the correction coefficient updating unit 17, and the transfer function filter C ^ _lm provided in the preceding stage constitute a correction coefficient updating unit. It

In each of the above embodiments, the active noise control system according to the present invention is applied to the vehicle active noise control system 1 for reducing the muffled noise transmitted from the engine 4 into the vehicle interior 6.
However, the application target of the present invention is not limited to this, for example, a device for reducing noise other than muffled noise such as road noise, or a vehicle interior such as a normal interior. A device or the like for reducing noise in a space other than 3 may be used.

The object of reduction is not limited to noise. For example, an engine mount (control vibration generating means) that generates an active control force is interposed between the engine 4 and a member, and the member side An acceleration sensor (residual vibration detection means) for detecting residual vibration is installed in
Then, if the engine mount is controlled based on the reference signal x and the output signal (residual vibration signal) of the acceleration sensor similar to those in the above-described embodiment, the vibration for the vehicle which can reduce the vibration transmitted from the engine 4 to the member side. It becomes a mold vibration control device.

Further, in each of the above embodiments, the so-called Filt is used as the update algorithm of the adaptive digital filter.
The case where the ered-X LMS algorithm is applied has been described, but the present invention is not limited to this. For example, the synchronous Filtered-X LMS algorithm (The Acoustical Society of Japan, Proceedings of the Acoustical Society of Japan, March 1992, 515-5).
See page 16 for details. ) And other algorithms.

[0063]

As described above, according to the present invention,
Since the output of the adaptive digital filter is not directly supplied as a drive signal to the control sound source or the controlled vibration source, the drive signal corrected based on the theoretically derived formula is supplied to the control sound source or the controlled vibration source. There is an effect that the harmonic distortion included in the control sound or the control vibration can be removed without using a particularly complicated structure.

Particularly, in the invention according to claim 2, 4, 6 or 8, since the correction coefficient is adaptively updated, the harmonic distortion included in the control sound or the control vibration can be surely removed. The effect is that you can do it.

[Brief description of drawings]

FIG. 1 is a block diagram showing a basic configuration of an invention according to claim 1.

FIG. 2 is a block diagram showing a basic configuration of an invention according to claim 3;

FIG. 3 is a diagram showing an overall configuration of a first embodiment.

FIG. 4 is a block diagram showing a functional configuration of a controller according to the first embodiment.

FIG. 5 is a flowchart showing an outline of processing executed in the controller.

FIG. 6 is a block diagram showing a configuration of simulation.

FIG. 7 is a diagram showing a result of simulation.

FIG. 8 is a diagram showing a result of simulation.

FIG. 9 is a block diagram showing a functional configuration of a controller according to a second embodiment.

FIG. 10 is a cross-sectional view showing the structure of a general speaker.

11 is a partially enlarged view of FIG.

FIG. 12 is an explanatory diagram of odd-order harmonic distortion.

FIG. 13 is an explanatory diagram of even-order harmonic distortion.

[Explanation of symbols]

 1 Vehicle Active Noise Control Device 2 Vehicle 3 Vehicle Room (Space) 4 Engine (Noise Source) 5 Crank Angle Sensors 8a, 8b Microphones (Residual Noise Detection Means) 9 Loudspeaker (Control Sound Source) 10 Controller 11 Reference Signal Generation Unit 12 Drive Signal Correction Section 15 Filter Coefficient Update Section 16 and 17 Correction Coefficient Update Section

─────────────────────────────────────────────────── ─── Continuation of the front page (51) Int.Cl. ⁶ Identification code Office reference number FI technical display location H03H 21/00 8842-5J

Claims (8)

[Claims] 1. A control sound source capable of generating a control sound in a space in which noise is transmitted from a noise source, and a reference signal generation means for detecting a noise generation state of the noise source and outputting it as a reference signal.
Residual noise detecting means for detecting residual noise at a predetermined position in the space and outputting it as a residual noise signal, and an adaptive digital filter with a variable filter coefficient for filtering the reference signal to generate a drive signal for driving the control sound source. And an adaptive processing means for updating the filter coefficient of the adaptive digital filter so that noise in the space is reduced based on the reference signal and the residual noise signal, the active noise control device comprising: An active noise control device comprising drive signal correction means for correcting a drive signal before being supplied to a sound source based on the following equation. _{y '= y + a 3 ×} y 3 , however, y' is the driving signal corrected, the drive signal before y is being corrected, a ₃ is the correction factor. 2. The active noise control device according to claim 1, further comprising correction coefficient updating means for updating the correction coefficient a ₃ based on the following equation. a ₃ (n + 1) = ν ₃ × a ₃ (n) −μ ₃ × e (n) ×
t ₃ where ν ₃ is the divergence suppression coefficient, μ ₃ is the convergence coefficient, e is the residual noise signal, and the terms with (n) and (n + 1) are values at discrete times n and n + 1, respectively. ing. Further, t ₃ is a value obtained by a convolution operation defined by the following equation, and C ^ _i is a transfer function obtained by modeling the transfer function C between the control sound source and the residual noise detecting means in the form of a finite impulse response function. The i-th filter coefficient of the filter C ^ and J are the number of taps of the transfer function filter C ^. 3. A control sound source capable of generating a control sound in a space where the noise is transmitted from the noise source, and a reference signal generating means for detecting a noise generation state of the noise source and outputting it as a reference signal.
Residual noise detecting means for detecting residual noise at a predetermined position in the space and outputting it as a residual noise signal, and an adaptive digital filter with a variable filter coefficient for filtering the reference signal to generate a drive signal for driving the control sound source. And an adaptive processing means for updating the filter coefficient of the adaptive digital filter so that noise in the space is reduced based on the reference signal and the residual noise signal, the active noise control device comprising: An active noise control device comprising drive signal correction means for correcting a drive signal before being supplied to a sound source based on the following equation. _{y '= y + a 2 ×} y 2 + a 3 × y 3 , however, y' is the driving signal corrected, the drive signal before y is being corrected, a _2, a ₃ is the correction factor. 4. The active noise control device according to claim 1, further comprising correction coefficient updating means for updating the correction coefficients a ₂ and a ₃ based on the following equation. a ₂ (n + 1) = ν ₂ × a ₂ (n) −μ ₂ × e (n) ×
t ₂ a ₃ (n + 1) = ν ₃ × a ₃ (n) −μ ₃ × e (n) ×
t ₃ where ν ₂ and ν ₃ are divergence suppression coefficients, μ ₂ and μ ₃ are convergence coefficients, and e is a residual noise signal, and (n) and (n + 1)
The terms with a mark represent values at discrete times n and n + 1, respectively. Further, t ₂ and t ₃ are values obtained by the convolution operation defined by the following equation, and C ^ _i
Is the i-th filter coefficient of the transfer function filter C ^ obtained by modeling the transfer function C between the control sound source and the residual noise detecting means in the form of a finite impulse response function, and J is the number of taps of the transfer function filter C ^. 5. A control vibration source capable of generating a control vibration that interferes with a vibration emitted from a vibration source, a reference signal generating means for detecting a vibration generation state of the vibration source and outputting it as a reference signal, the interference. A residual vibration detecting means for detecting a residual vibration afterwards and outputting it as a residual vibration signal; an adaptive digital filter having a variable filter coefficient for filtering the reference signal to generate a drive signal for driving the controlled vibration source;
An active vibration control device comprising: adaptive processing means for updating the filter coefficient of the adaptive digital filter so as to reduce the vibration after the interference based on the reference signal and the residual vibration signal. An active vibration control device comprising drive signal correction means for correcting a drive signal before being supplied to the drive circuit according to the following equation. _{y '= y + a 3 ×} y 3 , however, y' is the driving signal corrected, the drive signal before y is being corrected, a ₃ is the correction factor. 6. The active vibration control device according to claim 5, further comprising correction coefficient updating means for updating the correction coefficient a ₃ based on the following equation. a ₃ (n + 1) = ν ₃ × a ₃ (n) −μ ₃ × e (n) ×
t ₃ where ν ₃ is the divergence suppression coefficient, μ ₃ is the convergence coefficient, e is the residual vibration signal, and the terms with (n) and (n + 1) are the values at discrete times n and n + 1, respectively. ing. Further, t ₃ is a value obtained by a convolution operation defined by the following equation, and C ^ _i is a transfer obtained by modeling the transfer function C between the control vibration source and the residual vibration detecting means in the form of a finite impulse response function. The i-th filter coefficient of the function filter C ^, and J are the number of taps of the transfer function filter C ^. 7. A control vibration source capable of generating a control vibration that interferes with a vibration emitted from a vibration source, a reference signal generating means for detecting a vibration generation state of the vibration source and outputting it as a reference signal, the interference. A residual vibration detecting means for detecting a residual vibration afterwards and outputting it as a residual vibration signal; an adaptive digital filter having a variable filter coefficient for filtering the reference signal to generate a drive signal for driving the controlled vibration source;
An active vibration control device comprising: adaptive processing means for updating the filter coefficient of the adaptive digital filter so as to reduce the vibration after the interference based on the reference signal and the residual vibration signal. An active vibration control device comprising drive signal correction means for correcting a drive signal before being supplied to the drive circuit according to the following equation. _{y '= y + a 2 ×} y 2 + a 3 × y 3 , however, y' is the driving signal corrected, the drive signal before y is being corrected, a _2, a ₃ is the correction factor. 8. The active vibration control device according to claim 7, further comprising correction coefficient updating means for updating the correction coefficients a ₂ and a ₃ based on the following equation. a ₂ (n + 1) = ν ₂ × a ₂ (n) −μ ₂ × e (n) ×
t ₂ a ₃ (n + 1) = ν ₃ × a ₃ (n) −μ ₃ × e (n) ×
t ₃ where ν ₂ and ν ₃ are divergence suppression coefficients, μ ₂ and μ ₃ are convergence coefficients, and e is a residual vibration signal, and (n) and (n + 1)
The terms with a mark represent values at discrete times n and n + 1, respectively. Further, t ₂ and t ₃ are values obtained by the convolution operation defined by the following equation, and C ^ _i
Is the i-th filter coefficient of the transfer function filter C ^ obtained by modeling the transfer function C between the controlled vibration source and the residual vibration detecting means in the form of a finite impulse response function, and J is the number of taps of the transfer function filter C ^.