JP3293922B2 - Active noise control device - Google Patents

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 for reducing noise by causing a control sound generated from a control sound source to interfere with noise transmitted from a noise source.
In particular, a transfer function between a loudspeaker that generates a control sound and a microphone that detects residual noise can be measured with a small amount of calculation.

[0002]

2. Description of the Related Art Conventional active noise control devices include those described in British Patent No. 2,149,614 and Japanese Patent Publication No. 1-501344. These conventional devices are, for example, noise reduction devices applied to a closed space such as a cabin of an aircraft, and microphones installed at a plurality of positions in such a closed space to detect sound pressure, and a microphone installed in the closed space. A plurality of loudspeakers that generate control sounds are provided, and based on the noise generation state of the noise source, the loudspeakers generate control sounds having a phase opposite to the noise transmitted to the closed space to cancel the noises.

[0003] As a method of generating a control sound emitted from a loudspeaker, PROCEEDINGS OF THE IEEE, VOL.
63 PAGE 1692, 1975, “ADAPTIVE NOISE CANSELLATION:
PRINCIPLES AND APPLICATIONS ”
DROW LMS 'algorithm is applied to multiple channels. The contents are described in a paper by the inventor of the above patent, “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 suitable algorithms for updating the filter coefficient of the adaptive digital filter.
In the ered-X LMS algorithm, a filter representing a transfer function from a loudspeaker to a microphone is set for all combinations of loudspeakers and microphones, and a reference signal representing a noise generation state of a noise source is set by the filter. Based on the processed value and the residual noise detected by each microphone, the filter coefficient of the adaptive digital filter provided for each loudspeaker is updated.

Here, in such an active noise control device, it is premised that a filter representing a transfer function from a loudspeaker to a microphone accurately represents the transfer function. Function and
When the deviation from the transfer function of the actual physical space is large, not only does the noise reduction effect decrease, but also the noise reduction effect is reduced by 9% in the frequency domain.
Conversely, when a phase difference close to 0 degree occurs, the light may diverge.

As a conventional technique for solving such a problem, there is a technique disclosed in Japanese Patent Application Laid-Open No. 3-259722, which discloses a noise generated in a compressor of a refrigerator and radiated outside through a machine room duct. Loudspeaker and a microphone for noise control in the machine room duct, and generate a control sound from the loudspeaker according to the driving state of the compressor. Each time the compressor is stopped, white noise is generated as an identification sound to measure the transfer function between the loudspeaker and the microphone, and the filter is identified, so that the noise control characteristics are not degraded while the compressor is stopped. ing.

[0007]

However, when the technology relating to the refrigerator described above is considered to be applied to a vehicle or the like as it is, the transfer function in the passenger compartment is varied depending on the temperature, humidity, opening and closing of windows, number of occupants, and the like. For example, even if the filter coefficient is updated every time the engine is stopped, the difference between the transfer function represented by the filter and the transfer function in the actual physical space is time-dependent. The noise increases with the lapse of time, and good noise control cannot be performed. That is, when the transfer function varies greatly, such as in a vehicle, it is desirable to always identify the filter by measuring the transfer function in parallel with the noise control.

Further, in the above-mentioned conventional technique, since white noise is used for measuring a transfer function, an arithmetic processing for generating white noise or a storage area for storing white noise is required. When a transfer function is measured based on noise, it is necessary not only to execute a fast Fourier transform (FFT) or an inverse Fourier transform (IFFT), but also to perform a convolution operation between an identification filter and white noise. Therefore, an expensive operation element capable of high-speed operation is required to enable constant measurement of the transfer function, which causes a cost increase.

The present invention has been made in view of such an unsolved problem of the prior art, and has a transfer function between a loudspeaker for generating a control sound and a microphone for detecting a residual noise. It is an object of the present invention to provide an active noise control device capable of preventing deterioration of control characteristics even when the transfer function fluctuates in a short time and greatly by making it possible to measure the noise with a small amount of calculation.

[0010]

According to one aspect of the present invention, there is provided a control sound source capable of generating a control sound in a space where noise is transmitted from a noise source, and a noise source of the noise source. A noise generation state detection means for detecting a generation state and outputting it as a reference signal, a residual noise detection 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 having a variable filter coefficient;
A drive signal generating unit that generates a drive signal of the control sound source by filtering the reference signal with the adaptive digital filter, a transfer function filter that models a transfer function between the control sound source and the residual noise detection unit, Adaptive processing means for updating a filter coefficient of the adaptive digital filter so as to reduce noise in the space based on a result of filtering the reference signal by the transfer function filter and the residual noise signal. In the active noise control device, a signal addition unit that can temporarily add a pulse signal synchronized with the reference signal to the drive signal, and two identification filters that have variable filter coefficients and one transfer function. And setting the one identification filter based on the residual noise signal when the signal addition means has performed the addition, and Identification filter setting means for setting the other identification filter based on the residual noise signal when the signal addition means does not perform the addition, and a filter coefficient of the one identification filter and a filter coefficient of the other identification filter Transfer function filter setting means for setting the transfer function filter based on a difference between filter coefficients of the filter.

According to another aspect of the present invention, a control sound source capable of generating a control sound in a space to which noise is transmitted from a noise source and a noise generation state of the noise source are detected. A noise generation state detecting means for outputting a reference signal, a residual noise detecting means for detecting a residual noise at a predetermined position in the space and outputting the residual noise signal, an adaptive digital filter having a variable filter coefficient, and the reference signal. Drive signal generating means for generating a drive signal of the control sound source by performing a filtering process with the adaptive digital filter;
A transfer function filter that models a transfer function between the control sound source and the residual noise detection unit; and a noise in the space based on a result of filtering the reference signal with the transfer function filter and the residual noise signal. An adaptive processing means for updating a filter coefficient of the adaptive digital filter so as to reduce the number of the adaptive digital filters, wherein a signal addition capable of temporarily adding a pulse signal synchronized with the reference signal to the drive signal is provided. Means, a filter coefficient variable and two identification filters for one of the transfer functions, and an average value obtained by synchronizing the residual noise signal when the signal adding means performs the addition with the one The filter average of the remaining noise signal when the signal addition means does not perform the addition is used as the filter coefficient of the identification filter of Identification filter setting means as a filter coefficient of the identification filter; and transfer function filter setting for setting the transfer function filter based on a difference between a filter coefficient of the one identification filter and a filter coefficient of the other identification filter. Means.

[0012]

According to the invention described in claim 1 or 2,
The drive signal generating means generates a drive signal for the control sound source by filtering the reference signal representing the noise generation state of the noise source with an adaptive digital filter. Generates a control sound that is correlated with the noise transmitted to the space.However, immediately after the control starts, the filter coefficient of the adaptive digital filter does not always converge to the optimum value. Do not interfere with each other, it cannot be said that noise in the space is reduced.

However, when the adaptive processing means updates the filter coefficients of the adaptive digital filter based on the result of filtering the reference signal with the transfer function filter and the residual noise signal so as to reduce the noise in the space, the control is performed. As the filter coefficient of the adaptive digital filter converges toward the optimum value as it progresses, the control sound and the noise interfere with each other to reduce the noise in the space.

Further, since the signal adding means temporarily adds a pulse signal synchronized with the reference signal to the drive signal, when the addition is performed, the drive signal of the control sound source is obtained by converting the reference signal to an adaptive digital signal. When the result of filtering by the filter and the pulse signal synchronized with the reference signal are superimposed, and the addition is not performed, the drive signal of the control sound source is the result of filtering the reference signal by the adaptive digital filter. It becomes itself.

[0015] According to the first aspect of the present invention,
The identification filter setting means sets one identification filter C ＾ ₁ based on the residual noise signal when the addition is performed, and sets the other identification filter C ＾ ₁ when the addition is not performed. The filter C ＾ ₀ is set based on the residual noise signal. Then, a difference is generated between the _one identification filter C # ₁ and the other identification filter C # _{0 by} an amount corresponding to the generation of the pulse signal. The component caused by the pulse signal included in the residual noise signal is an impulse response of a transfer function between the control sound source and the residual noise detecting means, assuming that the magnitude of the pulse signal is “1”.

Therefore, the transfer function filter is set by the transfer function filter setting means based on the difference between the filter coefficient of the identification filter C # _{1 and} the filter coefficient of the identification filter C # ₀ . On the other hand, in the invention according to claim 2, the identification filter setting means synchronizes the filter coefficient of _one identification filter C ＾ ₁ with the residual noise signal when the addition is performed. When the above addition is not performed, the filter coefficient of the other identification filter C ＾ ₀ is set to an average value synchronized with the residual noise signal when the addition is not performed. . The method of calculating these average values is not particularly limited, and the average value may be calculated from the cumulative total within a predetermined time, or the moving average may be obtained. Alternatively, an algorithm that converges to a synchronized average value using the LMS algorithm may be used.

Here, the noise generated by the noise source is X, the reference signal is x, the pulse signal synchronized with the reference signal is x ', the residual noise is E, and the transfer function between the noise source and the residual noise detecting means is X. G, the adaptive digital filter is W, and the transfer function between the control sound source and the residual noise detection means is C. If the addition is performed and the drive signal of the control sound source is the superimposed signal, the residual The noise E is represented by the following equation (1).

E = GX + CWx + Cx ′ (1) On the other hand, if the addition is not performed and the drive signal is the result of filtering the reference signal with the adaptive digital filter, the pulse signal x ′ is not generated. , The residual noise E is expressed by the following equation (2). E = GX + CWx ...... (2 ) Then, the filter coefficient of the identifying filter C ^ ₁ is the average value taken synchronization residual noise E represented by the above formula (1), of the identifying filter C ^ ₀ Each filter coefficient is an average value obtained by synchronizing the residual noise E expressed by the above equation (2). However, since each of the filter coefficients is not an instantaneous value but an average value obtained by adjusting the starting points in time, Identification filter C ＾ ₁
And a noise filter C た め₀
If the difference between the filter coefficient of the identification filter C # _{1 and} the filter coefficient of the identification filter C # ₀ is determined even if the residual noise signal for calculating The first and second terms on the right side of the equation (1) and the first and second terms on the right side of the equation (2) are canceled, and a result as shown in the following equation (3) is obtained.

C ＾ ₁ −C ＾ ₀ = Cx ′ (3) That is, the transfer function filter setting means sets the filter coefficient of the identification filter C ＾ ₁ and the identification filter C set by the identification filter setting means. When the difference between the filter coefficients of ＾ ₀ is obtained, as is apparent from the above equation (3), the transfer function C between the control sound source and the residual noise detection means in the actual physical space is added to the pulse synchronized with the reference signal x. A response Cx 'to which the signal x' has been applied is obtained.

Therefore, for example, when the pulse signal x ′ is
If the pulse signal (impulse signal) is (3)
Since the result of the expression represents the impulse response of the transfer function C, the transfer function filter can be easily and accurately set by the transfer function filter setting means.

[0021]

Embodiments of the present invention will be described below with reference to the drawings. FIG. 1 is a diagram showing an overall configuration of a first embodiment of the present invention. This embodiment is directed to a vehicle for reducing a muffled sound transmitted from an engine 4 as a noise source to a vehicle interior 6 as a space. The present invention is applied to an active noise control device 1 for use.

First, the construction will be described. The vehicle body 3 is supported by suspensions interposed between the vehicle body 3 and the front wheels 2a, 2b, the rear wheels 2c, 2d and the wheels 2a to 2d. The vehicle shown in FIG. 1 is a so-called front-engine front-wheel drive vehicle in which front wheels 2a and 2b are rotationally driven by an engine 4 arranged in the front part of the vehicle body 3. A crank angle sensor 5 is attached to the engine 4,
The crank angle sensor 5 supplies the controller 10 with a crank angle signal X synchronized with the rotation of the crank angle of the engine 4 (for example, one pulse signal every time the crank rotates 1 °).

In the cabin 6 of the vehicle body 3, loudspeakers 7a, 7b, 7c and 7d as control sound sources face the front seats S ₁ and S ₂ and the rear seats S ₃ and S ₄ respectively. It is arranged in the door part. In addition, each seat S ₁
The head restraint position of the to S ₄ is a microphone 8a~8h as residual noise detecting means, have been respectively disposed two by two, the residual noise signal e ₁ to e these microphones 8a~8h was measured as a sound pressure ₈ Is supplied to the controller 10.

The controller 10 includes a microcomputer, necessary interface circuits, and the like, and includes a crank angle signal X supplied from the crank angle sensor 5 and a residual noise signal supplied from the microphones 8a to 8h. Based on e _{1 to} e ₈ , the loudspeakers 7a perform arithmetic processing described later so that control sounds that cancel the muffled sound transmitted into the vehicle interior 6 are emitted from the loudspeakers 7a to 7d. and outputs a drive signal y ₁ ~y ₄ to ～7D.

FIG. 2 is a block diagram showing a functional configuration of the controller 10. The controller 10 has a cycle (for example, the same cycle as that of vibration generated in the engine 4 causing a muffled sound, based on the crank angle signal X). In the case of a four-cylinder reciprocating cylinder, a reference signal generating unit 1 for generating and outputting a reference signal x formed of a sine wave (180 degrees of the crank is defined as one cycle)
1 and a pulse signal generator 12 for generating and outputting a pulse signal x 'having the same cycle as the reference signal x based on the reference signal x.
And

The controller 10 has a number corresponding to the loudspeakers 7a to 7d (M: in this embodiment, M
= 4) adaptive digital filter W _m (m = 1 to M)
If, each filter coefficient of the adaptive digital filter _{W m W mi (i = 0~I} -1: I adaptive digital filter W
A drive signal generation unit 13 as drive signal generation means for generating and outputting drive signals y _{1 to} y ₄ by convolving the number of taps of _m (the number of filter coefficients) and the reference signal x; A transfer function filter C 関 数_lm (l = 1 to L: L is the number of the microphones 8 a to 8 h, in which the transfer function between the microphones 7 a and 8 a to 8 h is modeled in the form of a finite impulse response function. = 8)
And each filter coefficient C _係数 of the transfer function filter C ＾ _lm
lmj (j = 0 to J-1: J is the number of taps of the transfer function filter C ＾ _lm ) and the reference signal x to convolve the reference processed signal r
a reference processed signal generating unit 14 for generating and _lm output, the reference processed signal r _lm and muffled sound in the passenger compartment 6 based on the residual noise signal e ₁ to e ₈ is in the drive signal generating unit 13 so as to reduce An adaptive processing unit 15 as an adaptive processing means for updating each filter coefficient W _mi of the adaptive digital filter W _m .

Furthermore, the controller 10 includes an adder capable of adding unit 16 into a pulse signal x 'the drive signal y _m that is generated by the pulse signal generator 12, the pulse signal generator 1
2 and an intermittent switch section 17 interposed between the adder section 16 and intermittently supplying the pulse signal x 'to the adder section 16, and two pulse signal x' provided for one transfer function filter C ＾ _lm. There filter identifying the filter coefficient variable one by one output in synchronization with the clock pulses C _P to be described later, from the time it is entered the filter coefficients in the order _C ^ 1lm, C ^ 0lm
When, a subtracting unit 18 for calculating a difference between the output of the residual noise signal e _l and identifying filter C ^ _{1 lm} or identifying filter _C ^ 0lm, identifying filter C ^ _{1 lm,} and a _C ^ 0lm subtraction section 18 And an identification filter C _{1lm , C _}
A change-over switch 19 for supply to the subtraction unit 18 to either the output of _0Lm, identifying filter C ^ _{1 lm,} each filter coefficient of the _{C ^ 0lm C ^ 1lmj, C} ^ 0lmj matches the output of the subtraction section 18 A filter coefficient updating unit 20 for updating the filter coefficients C ＾ _1lmj and C ＾ _0lmj as _{described above} , and a filter coefficient updating unit 20 interposed between the filter coefficient updating unit 20 and the identification filters C ＾ _1lm , C ＾ _0lm and the identification filters C ＾ _1lm , The changeover switch unit 21 that switches so that one of the filter coefficients of C ＾ _0lm is updated, and the difference between the filter coefficients C ＾ _1lmj and C ＾ _0lmj of the identification filters C ＾ _1lm and C ＾ _0lm. And a subtraction unit 22 for setting a transfer function filter C ＾ _lm .

However, the intermittent switch section 17, the changeover switch section 19 and the changeover switch section 21 are interlocked with each other, and when the intermittent switch section 17 is in the connected state, the changeover switches 19 and 21 are connected to the identification filter C ＾ _1lm. Side, and when the intermittent switch 17 is in the cut-off state, the changeover switches 19 and 21 are switched to the identification filter C _{# 0lm} side.

Further, based on the crank angle signal X, the controller 10 controls the frequency of the reference signal x to be N (N is an integer) times the frequency (for example, in the case of a reciprocating four-cylinder engine, if N = 10, the crank is 18 times). degrees consisting of rotating every a single pulse) has a clock pulse generator 23 to the clock pulse C _P to produce an output, each processing in the controller 10 is basically synchronized with the clock pulses C _P And executed. Therefore, the adaptive digital filter W _m and the identification filters C ＾ _1lm and C ＾ _0lm output N times per cycle of the reference signal x having the same cycle as the muffled sound.

Here, in the present embodiment, the filter coefficient updating unit 15 is an LM which is one of the algorithms suitable for updating the filter coefficient of the adaptive digital filter.
Based on the S algorithm, the adaptive digital filter W
_m , each filter coefficient W _mi is updated, but in particular the reference signal x
Is filtered using a transfer function filter C ＾ _lm , the filtered r- _lm algorithm is executed, and the updating equation of the filter coefficient W _mi of the adaptive digital filter W _m is (4)
Equation or equation (5) is obtained.

[0031] λ _W is a divergence suppression coefficient and takes a value of 1 or less. Α _W is a coefficient called a convergence coefficient, and is related to the speed at which the filter converges optimally and its stability. In addition,
The term to which (n) is attached indicates that it is a value at the sampling time n, and the subscript k is a variable representing the number of processing in one cycle (N times), To N-1.

On the other hand, the identification filters C ＾ _1lm , C ＾ _0lm
The filter coefficient updating unit 20 for updating the filter coefficient of
In the present embodiment, the LMS
Identifying filter so that the output of the subtracting unit 18 based on the algorithm becomes zero C ^ _{1 lm,} each filter coefficient of the _C ^ 0lm C ^ 1lmj, although to update the _C ^ 0lmj, pulse signal x 'is the filter coefficient Since the data is input to the updating unit 20, a so-called synchronous LMS algorithm is executed. Note that the synchronous LMS algorithm is an LMS algorithm using a pulse signal synchronized with the fundamental frequency of noise as a reference signal (see “Acoustic Society of Japan, March 1994, March 199”, pp. 515-516). .).

Therefore, the identification filter C ＾_1lm, C ＾
_0lmEach filter coefficient C ＾_1lmj, C ＾ _0lmjThe update formula for
The following equation (6) is obtained. C ＾_{* lmk}(N + 1) = λ_CC ＾_{* lmk}(N) -α_Ce_l …… (6) where λ_CIs the divergence suppression coefficient, α_CIs the convergence coefficient,
* Is 1 or 0. Note that this equation (6) is
Filter C ＾_1lm, C ＾_0lmIs less than or equal to N
The tap number J is larger than N
In this case, the following equation (7) is used instead of equation (6).
Will be.

[0034] _{C ^ * lm (k + Np} ) (n + 1) = λ C C ^ * lm (k + Np) (n) -α C e l ...... (7) the case of using the equation (7), p , k is the quotient and remainder when the number of clock pulses C _P of the pulse signal x 'has counted from the time it is added to the drive signal y _m at the addition section 16 is divided by N. That is, the above equation (6) is equivalent to the above (7)
In the equation, it can be considered that p = 0.

FIG. 3 is a flowchart showing the outline of the processing executed in the controller 10. The operation of this embodiment will be described below. That is, processing in the controller 10 in is not so the process of synchronization to one sampling clock pulses C _P as described above is executed,
First, in step 101, the residual noise signal e _l
Reading, then in step 102, and the reference signal x filtered by a transfer function filter C ^ _lm calculates a reference processed signal r _lm.

[0036] Then, the process proceeds to step 103, in accordance with the above (4) or (5), to update the filter coefficients W _mi of the adaptive digital filter W _m. Step 103
After updating all the filter coefficients W _mi in step, the process proceeds to step 104, and it is determined whether or not the flag F ₁ is 1. The flag F _1, if either of the drive signal y ₁ ~y ₄ loudspeaker 7a to 7d, when adding the serving identification signal pulse signals x 'to (F ₁ = ₁₎ or not added (F ₁ = 0 ), And is appropriately set in other processing described later.

[0037] If the determination in step 104 is "NO", since any identification signal to the drive signal y _m can be judged not to be added, the process proceeds to step 105, the filter coefficient of the identifying filter _C ^ 0lm C ＾ _{0lm (k + Np)} is updated according to the above equation (7). In this case, since the pulse signal x 'is input to the filter coefficient updating unit 20 as the reference signal, it is not necessary to update all the filter coefficients C ＾ _0lmj, and only the k + Np-th coefficient.

On the other hand, the determination in step 104 is "YES".
For the either it determines that pulse signal x 'is added to the drive signal y _m, the process proceeds to step 106. In step 106, the value of the flag F ₂ is determines whether or not 1, the flag F ₂ is one of the drive signal y
A _{flag 1} ~y ₄ to the pulse signal x 'represents whether the summed, in this embodiment, four loudspeakers 7
since it has A～7d, flag F ₂ takes a value of 1-4.

Therefore, steps 106, 107, 108
If the flag F ₂ is equal to or 1, 2, 3, the determination in step 106 of "YES" in (F ₂ =
In step 1), the process proceeds to step 109. If the determination in step 107 is “YES” (F ₂ = 2), step 110 is executed.
When the determination in step 108 is “YES” (F ₂ = 3), the process proceeds to step 111,
If the determination at 08 is “NO” (F ₂ = 4), the process proceeds to step 112.

In each of the processes of steps 109 to 112, the identification filter C is calculated according to the above equation (7).
The filter coefficients C ＾ _{1lm (k + Np)} of ＾ _1lm and C ＾ _0lm are updated. In step 109, m = 1 and step 110
M = 2, step 111, m = 3, step 11
In 2, the above equation (7) is updated with m = 4. This is because the identification filters C ＾ _1lm and C ＾ _0lm are updated in order to identify the transfer function filter C 、 _lm , but the loudspeakers 7a to 7d and the microphone 8a
In order to measure the transfer function between the loudspeakers 7a to 7h, if the identification sounds are generated from the loudspeakers 7a to 7d, the loudspeakers 7a
Since the identification sounds emitted from the speakers 7 to 7d arrive at the same time and an accurate transfer function cannot be measured, the identification sound is generated only from one loudspeaker 7a to 7d at a time as described later. This is for correspondence.

Update processing of step 105 or step 1
After completing any of the update processing of 09 to 112, and proceeds to step 113, by subtracting the respective filter coefficients _C ^ 0lmj of identifying filters _C ^ 0lm from each filter coefficient _C ^ 1lmj of identifying filters C ^ _{1 lm} results , A transfer function filter C ＾ _lm is set. The reason why the transfer function filter C ＾ _lm can be set based on such an operation will be described later.

In step 113, the transfer function filter C ＾ _lm
After making the settings, the process proceeds to step 114, to zero clear all of the drive signal y ₁ ~y _4. And step 1
Then, it is determined whether or not the variable k has reached N. If it is determined that the variable k has not yet reached N, step 1 is executed.
The process proceeds to 16 to increment the variable k. On the other hand, if the determination in step 115 is “YES”, it means that the variable incremented by one in step 116 reaches N each time the clock pulse _CP is generated.
In other words, the current time point is a point in time when one cycle of the reference signal x has passed and a new cycle has started.

Then, the process proceeds to step 117 to clear the variable k for counting the number of processes in one cycle to zero, and then proceeds to step 118 to determine whether the variable p has reached a constant P (P is an integer). Is determined. Here, the variable p is the quotient when the number of clock pulses C _P which is counted from the time when the pulse signal x 'to the drive signal y _m is added as described above divided by N, and more specifically, from the time the pulse signal x 'is added to the drive signal y _m, which indicates whether the period of the reference signal x has passed many. The constant P is an integer indicating how many times the cycle number N of the tap J of the transfer function filter C ＾ _lm is set, and is set in advance in consideration of the reverberation time in the vehicle interior. For example, to make the number of taps J 2.5 times the period N, the constant P is set to 3.

[0044] If step 118 is "NO", yet elapsed time corresponding to the pulse signal x 'to one of the drive signal y _m still number of taps from the time that caused the identified sound is added to J is More specifically, since it can be determined that the updating process of the identification filter in steps 105 and 109 to 112 has not been completed, if a new identification sound is generated, accurate identification of the transfer function will be performed. Is determined to be disturbed, the process proceeds to step 119, and the variable p is incremented.

However, the determination at step 118 is "YE
In the case of "S", it can be determined that the update processing of the identification filter in steps 105 and 109 to 112 has been completed, so the processing shifts to step 120 to clear the variable p to zero, and then executes the processing of step 121 and thereafter. Step 1
In 21, the flag F ₁ is determined whether or not 1. Here, if it is determined that the flag F ₁ is not 1, the process of step 105 is executed last time, and the identification filter C
Since ^ filter coefficient C ^ _0Lmj of _0lm it can be determined to have been updated, in the next processing, of identifying filters C ^ _{1 lm,} filter coefficients of the identification filter C ^ _1L1 of m = 1 C
_{フ ラ グ} Set a flag to update _1l1j . Specifically, sets the flag F ₁ to 1 proceeds to step 122, the flag F ₂ proceeds to step 123 1
Then, the process proceeds to step 124 to set the drive signal y ₁ of m = ₁ to a predetermined value b. This predetermined value b is
This is a constant that is regarded as a superimposition of an impulse of magnitude 1 on the drive signal y _m , and in FIG. 2 corresponds to the pulse signal x ′.

[0046] When the flag F ₁ in step 121 is determined to be 1, since the last time can be determined with any of the update process of step 109 to 112 is executed, the process proceeds to step 125, the flag F ₂ is It is determined whether it is 1 or not. When the determination in step 125 is “NO”, the process proceeds to step 126 to determine whether or not the flag F ₂ is 2, and when the determination in step 126 is “NO”, the process proceeds to step 127. it is determined whether the flag F ₂ is 3.

That is, the determination in step 125 is “YES”
In the case of, since the last time can be judged to processing in step 109 the filter coefficients _C ^ 1l1j of identifying filters C ^ _1L1 is executed is updated, in the next process, the update process corresponding to m = 2 is performed as is, sets the flag F ₂ to 2 proceeds to step 128, sets the driving signals y ₂ to a predetermined value b.

Then, the determination in step 126 is "YE
In the case of "S", m = 2 in the previous step in step 110
Is performed, and if the determination in step 127 is "YES", it can be determined that the update processing corresponding to m = 3 was performed in the previous step 111, so that the next time m = 3 , M = 4, the processes of steps 130 and 131 or steps 132 and 133 are executed.

If the determination in step 127 is "NO", it can be determined that the update processing corresponding to m = 4 was performed in step 112 last time. There was judged complete, the next time, as update processing at the step 105 is performed, the flag F ₁ proceeds to step 134 to set to zero. Then, steps 116, 119, 124, 129, 13
After executing the processing of 1, 133 or 134, step 1
The process proceeds to 35, where the drive signal y _m is calculated according to the following equation (8).

Y _m = y _m + x * W _m (8) Note that “*” on the right side of the above equation (8) represents a convolution integral. Here, the value y _{m of} the first term on the right side of the above equation (8) is obtained by executing steps 114 and 115 and 11
8 or 127 is determined in the case of "NO" is 0 for all of the drive signal y _m. Therefore, each of the drive signals y ₁ to y ₁
y ₄ is the reference signal x and the adaptive digital filter W
_m is a value obtained by convolving each filter coefficient W _mi with _m , and FIG.
In will be described, the output of the drive signal generator 13, as a driving signal y _m loudspeakers 7a to 7d.

However, steps 124, 129, 131
Or if the 133 either is executed, either the value of the drive signal y ₁ ~y ₄ is set to a predetermined value b, for the drive signal y _m that is not set to a predetermined value b , convolving the filter coefficients W _mi of the reference signal x and an adaptive digital filter W _m values loudspeaker 7a~
Although the 7d driving signal, for driving signals y _m which is set to a predetermined value b, and the respective filter coefficients W _mi of the reference signal x and an adaptive digital filter W _m to the value convolution,
(Will be described in FIG. 2, a value obtained by adding a pulse signal x 'in the addition unit 16 to an output y _m of the drive signal generating unit 13) further predetermined value b is added value is, loudspeakers 7a~7
the drive signals y _m of d.

[0052] After completing the process of step 135, the process proceeds to step 136, and outputs each of the driving signals y _m to each loudspeaker 7a to 7d. Then, the loudspeakers 7a to
Although control sound into the passenger compartment 6 from 7d is generated, since after start of control is not necessarily the filter coefficient W _mi of the adaptive digital filter W _m is converged to an optimal value, necessarily the passenger compartment 6 It cannot be said that the transmitted muffled sound is reduced.

However, when the processing shown in FIG. 3 is repeatedly executed, the filter coefficient W _mi is sequentially updated in step 103 according to the above equation (4) or (5) according to the LMS algorithm, and toward the optimum value. Since the sound converges, the muffled sound transmitted into the cabin 6 is transmitted to the loudspeakers 7 a to 7.
The sound is canceled by the control sound emitted from d, and the noise in the vehicle interior 6 is reduced.

In this embodiment, the drive signal y
the case of directly outputting _m, there is a case of outputting by adding a predetermined value b to any one of the drive signal y _m at a timing synchronized with the drive signal x, when outputting without the addition, the step In step 105, while updating the identification filter C ＾ _0lm in accordance with the above equation (7), when adding and outputting, in one of steps 109 to 112,
The identifying filter C ^ _{1 lm} corresponding to the summed drive signal y _m is set to be updated in accordance with equation (7).

Here, the above equation (7) is an updating equation based on the LMS algorithm.
The input to _1lm , C ＾ _0lm is a pulse signal x ′, and if the magnitude is considered to be 1, these identification filters C ′
The output of ＾ _1lm , C ＾ _0lm is the filter coefficient C
Since ＾ _1lmk and C ＾ _0lmk are _themselves , if the filter coefficients C ＾ _1lmk and C ＾ _0lmk are sequentially updated according to the above equation (7), the filter coefficients C ＾ _1lmk and C ＾ _{0lmk become} variable k
Average value of the residual noise signal e _l (average value synchronized)
Converges to

That is, when the pulse signal x ′ is not added, the identification filter C ＾ updated according to the above equation (7)
_0lm, for example, as shown in FIG. 4 (a), it will represent the waveform of the residual noise signal e _l at that time. Similarly, the identification filter C updated when the pulse signal x 'is added
^ _{1 lm} also the residual will represent the waveforms of the noise signal e _l at that time, in this case, since the identification sound corresponding to the pulse signal x 'is emitted from the loudspeaker with the control sound, residual noise signal e _l includes not only residual noise due to muffled sound and the like, but also identification sound emitted from the loudspeaker and reaching the microphones 8a to 8h.

Accordingly, the identification filter C ＾ _1lm has a slightly distorted waveform as compared with FIG. 4A _showing the shape of the identification filter C ＾ _0lm , for example, as shown in FIG. 4B. The distorted portion is caused by the identification sound. Therefore, each filter coefficient C _係数 of the identification filter C ＾ _{1lm
From 1lmj} , identification filter C ＾ _0lm each filter coefficient C ＾
_{When 0lmj} is subtracted, as shown in FIG. 4 (c), only a waveform due to the identification sound remains, and this waveform causes the loudspeaker to generate an identification sound corresponding to the pulse signal x '. Since this corresponds to the sound measured by the microphones 8a to 8h, it should match the impulse response of the transfer function between the loudspeakers and the microphones 8a to 8h.

[0058] Incidentally, the residual noise component other than the identified sound included in the identification filter C ^ _{1 lm,} identifying filter _C ^ 0lm
Are not instantaneously coincident with the residual noise components other than the identification sound included in the residual noise signal e _l because they are temporally shifted in their identification. Is the average value of the filters for identification C _{1lm , C _}
The algorithm is such that the filter coefficients C ＾ _1lmj and C ＾ _{0lmj of 0 lm} are obtained.
^ _{For 1lm} a process for updating and the identifying filter C ^ _{0l m} to frequently switch between process of updating, after appropriately repeating the process shown in FIG. 3, the residual noise component other than the identified sound matching It can be regarded as having done.

[0059] Therefore, in step 113, the transfer function filter C ^ _lm on the basis of the result of subtracting the respective filter coefficients _C ^ 0lmj of identifying filters _C ^ 0lm from each filter coefficient _C ^ 1lmj of identifying filters C ^ _{1 lm} If set, even if the transfer function between the loudspeakers 7a to 7d and the microphones 8a to 8h changes from the initial state according to, for example, changes in the temperature, humidity, window opening / closing status, and the number of occupants in the passenger compartment 6, The transfer function filter C ＾
_{Since lm} is changed, the difference between the transfer function in the actual physical space and the transfer function represented by the transfer function filter C ＾ _lm can be minimized, and a good noise reduction effect can be obtained. .

Moreover, the identification filters C _{1lm , C _}
0Lm also eventually higher the identification accuracy since it converges to the average value taken synchronization residual noise signal e _l, and, for those identified filter C ^ _{1 lm,} filter coefficients C ^ _0lm C ^
_Since the transfer function filter C ＾ _lm is identified by a simple operation of subtraction of _1lmj and C ＾ _0lmj , the transfer function filter C ＾ _lm can be set with high accuracy. This means that it is not necessary to increase the power of the identification sound so much, so that the generation of the identification sound does not give an occupant a discomfort.

In this embodiment, since the transfer function filter C ＾ _lm is identified based on the pulse signal x ′ instead of the white noise, the following advantages are provided. First, in order to generate white noise, it is necessary to generate a random signal having constant power over the entire frequency band. While the configuration in which the data is stored causes an increase in the memory capacity, the generation of the pulse signal x 'is very easy, so that the calculation load or the memory capacity can be reduced.

Second, an identification filter C _{1lm , C _}
Since the input to _0lm is a pulse signal x ', whereas it is necessary to perform convolution would otherwise need only outputs filter coefficient _C ^ 1lmj, a _C ^ 0lmj in synchronism with the clock pulses C _P Therefore, the calculation load can be greatly reduced. Third, in the present embodiment, the filter coefficients C ＾ _1lmj , C ＾ of the identification filters C ＾ _1lm , C ＾ _{0lm.
Although} the LMS algorithm is applied to update _{0lmj, since} the pulse signal x 'is applied as the reference signal, the k + Np-th filter coefficient C ＾ _{* lm (k Since} only _{+ Np)} needs to be updated, the calculation load can be reduced.

As described above, according to the configuration of the present embodiment, the transfer function filter _Clmlm can be identified with a very small amount of operation and a high accuracy compared with the conventional one, so that an expensive operation element capable of high-speed operation is provided. , And a good noise reduction effect can be obtained even in a space where the transfer function changes frequently, such as a vehicle, without introducing discomfort to the occupant.

Here, in this embodiment, the noise generation state detecting means is constituted by the crank angle sensor 5 and the reference signal generating section 11, and the adding section 16 and the intermittent switch section 17 are provided.
And steps 124,129,131,133,135
Constitutes a signal adding means, and the subtracting section 18,
The selection switch units 19 and 21, the filter coefficient update unit 20, and the processing of steps 105 to 112 constitute an identification filter setting unit, and the subtraction unit 22 and step 113
The transfer function filter setting means is constituted by the above processing.

FIG. 5 is a block diagram showing the configuration of the second embodiment of the present invention. As in FIG. 2 of the first embodiment, FIG.
2 shows a functional configuration of the controller 10. In this embodiment, similarly to the first embodiment, the present invention is applied to an active noise control device for a vehicle for reducing a muffled sound transmitted from an engine into a vehicle interior. The other configuration is the same as that of the first embodiment.

[0066] That is, this embodiment, synchronous LMS regard adaptive digital filter W _m to generate a drive signal y _m
Algorithm (more specifically, synchronous Filtere
d-X LMS algorithm).
Therefore, unlike the first embodiment, the reference signal generator 11 generates and outputs a reference signal x composed of an impulse train synchronized with the muffled sound, based on the crank angle signal X.

Then, the reference signal x is supplied to the drive signal generation unit 13, while the addition unit 16, the identification filters C _{# 1lm} and C _{# 0lm} are supplied as pulse signals synchronized with the reference signal x.
And a filter coefficient updating unit 20. further,
In the first embodiment to omit the reference processed signal generating unit 14 that existed, the filter coefficient W of the adaptive digital filter W _m
The output of the subtraction unit 30 that calculates the difference between the output of the identification filter C _{# 1lm and} the output of the identification filter C _{# 1lm} is supplied to the filter coefficient update unit 15 that updates _mi as it is. That is, in the present embodiment, the identification filters C _{# 1lm} and C # _{0lm also} have the function of the transfer function filter C # _lm in the first embodiment.

FIG. 6 is a flowchart showing the outline of the processing in the controller 10 of the present embodiment. The same reference numerals are given to the same processes as those of FIG. 3 described in the first embodiment, and the same explanation will be given. Is omitted. That is, in this embodiment, _after the filter coefficient C _{係数 0lm (k + Np)} of the identification filter C ＾ _0lm is updated in step 105, the _{process proceeds} to step 201, and according to the following equation (9). , The negative change amount Δr _lmk of the filter coefficient C ＾ _{0lm (k + Np)} is calculated.

Δr _lmk = − {C ＾ _{0lm (k + Np)} (n + 1) −C ＾ _{0lm (k + Np)} (n)｝ (9) Then, the _{process proceeds to} step 202 and the following (10) According to the equation, the reference processing signal r _lmk used for updating the filter coefficient W _mi of the adaptive digital filter W _m in step 103
Is accumulated.

R _lmk = r _lmk + Δr _lmk (10) On the other hand, after updating the filter coefficient C ＾ _{1lm (k + Np)} of the identification filter C ＾ _{1lm in} step 109, 110, 111 or 112, The process proceeds to steps 203, 205, 207 or 209, and calculates a change amount Δr _lmk of the filter coefficient C ＾ _{1lm (k + Np)} according to the following equation (11).

Δr _lmk = C ＾ _{0lm (k + Np)} (n + 1) −C ＾ _{0lm (k + Np)} (n) (11) Then, the _{process proceeds to} step 204, 206, 208 or 210, and According to equation (12), the reference processing signal r _lmk
Is accumulated. r _lmk = r _lmk + Δr _lmk (12) The reference processing signal r _lmk can be obtained by such an operation because the reference signal x is an impulse train, the input to the transfer function filter C ＾ _lm and the identification. Filter C
＾ _1lm and C ＾ 0 _lm are equal to the input, and both outputs are filter coefficients C ＾ _lmk , C ＾ _1lmk , and _{Clm 0 lmk} . Further, as described in the first embodiment, the transfer function filter C _{フ ィ ル タ lm} filter coefficient _{Clm lmj} is the identification filter C _{同 定
1 lm,} C ^ filter coefficient _0lm C ^ _1lmj, because of being determined from the difference between the _C ^ 0lmj, they filter coefficient C ^
_If the changes of _{1lmk and Clm0lmk} are accumulated with different polarities, each filter coefficient C ＾ _{lmj of the} transfer function filter C ＾ _lm
This is because the same result as that of calculating the reference processing signal r _lmk by _{convolving the} reference processing signal x with the reference signal x is obtained.

[0072] Further, as a result of the reference signal x and an impulse train, the operation of the drive signal y _m at step 135,
This is performed according to the following equation (13). _{y m = y m + W mk} ...... (13) Thus, with the configuration of this embodiment, there is an advantage that the further reduction of calculation load than the first embodiment can be achieved. Other functions and effects are the same as those of the first embodiment.

In this embodiment, as a result of reducing the calculation load, the transfer function filter C ＾ _lm does not exist in the form, and the identification filters C ＾ _1lm and C ＾ _0lm are used in the transfer function filter CＣ. Since it also has the function of ＾ _lm , the processing in the identification filters C ＾ _1lm , C ＾ _0lm , the subtraction unit 30, and steps 201 to 210 corresponds to the transfer function filter setting means.

In each of the above embodiments, the case where the active noise control device according to the present invention is applied to a device for reducing the muffled sound of a vehicle, which is a periodic noise, has been described. For example, the present invention is applicable to a device for reducing random noise such as road noise. That is, for example, if the device aims to reduce road noise,
The reference signal x is a vertical acceleration generated in the suspension. The reference signal x is processed by a low-pass filter that extracts a low frequency of about 30 to 40 Hz, and a pulse signal x 'is generated in synchronization with the low-frequency component. In this case, the same operation and effect as in the first embodiment can be obtained.

In each of the above embodiments, the case where the active noise control device according to the present invention is applied to a vehicle has been described. However, the application object of the present invention is not limited to this. Naturally, it can be applied to a space.
Furthermore, in the above embodiments, by utilizing the synchronous LMS algorithm, identifying filter C ^ _{1 lm,} each filter coefficient of the _C ^ 0lm is, as a mean value synchronizing the residual noise signal e _l However, the method of _obtaining the synchronized average value of the residual noise signal el is not limited to this.
For example, advance accumulates synchronize the residual noise signal e _l within a predetermined time may be calculated by dividing the accumulated result in accumulation times, or a moving average with a weighting You may use the technique of finding.

[0076] Furthermore, in the above embodiments, identifying filter C ^ _{1 lm,} it has been described for the case of setting the average value synchronized for each filter coefficient residual noise signal e _l of _C ^ 0lm, if it is clear that the noise remaining in the space is in the less stable state of change, the remaining noise signal e identifying filter the instantaneous value of _l C ^ _{1 lm,} even the filter coefficients of _C ^ 0lm, transfer function Is possible.

[0077]

As described above, according to the present invention,
Because two identification filters are provided for one transfer function, the identification filters are appropriately updated, and the transfer function filter is set based on the difference between the filter coefficients of the two identification filters. A space in which the transfer function changes frequently like a vehicle without introducing expensive operation elements capable of performing high-speed operation, which makes it possible to identify a transfer function filter with a very small amount of operation compared to the conventional art. However, there is an effect that a good noise reduction effect can be obtained.

[Brief description of the drawings]

FIG. 1 is a diagram showing an overall configuration of a first embodiment of the present invention.

FIG. 2 is a block diagram illustrating a functional configuration of a controller according to the first embodiment.

FIG. 3 is a flowchart illustrating an outline of processing executed in a controller according to the first embodiment;

FIG. 4 is a waveform chart for explaining the reason why a transfer function can be set.

FIG. 5 is a block diagram illustrating a functional configuration of a controller according to a second embodiment.

FIG. 6 is a flowchart illustrating an outline of processing executed in a controller according to a second embodiment.

[Description of Signs] 1 Active noise control device for vehicle 4 Engine (noise source) 5 Crank angle sensor 6 Cab (space) 7a to 7d Loudspeaker (control sound source) 8a to 8h Microphone (residual noise detection means) 10 Controller Reference Signs List 11 Reference signal generation unit 12 Pulse signal generation unit 13 Drive signal generation unit (drive signal generation unit) 14 Reference processing signal generation unit 15 Filter coefficient update unit (adaptive processing unit) 16 Addition unit 17 Intermittent switch unit 18, 22 Subtraction unit 19 , 21 changeover switch unit 20 the filter coefficient updating unit W _m adaptive digital filter C ^ _lm transfer function filter _C ^ 1lm, C ^ 0lm identifying filters

Continuation of the front page (56) References JP-A-62-193310 (JP, A) JP-A-4-267298 (JP, A) (58) Fields investigated (Int. Cl. ⁷ , DB name) G10K 11 / 178 B60R 11/02 F01N 1/00 F01N 1/06 H03H 17/00 601 H03H 17/02 601 H03H 21/00

Claims (2)

(57) [Claims] 1. A control sound source capable of generating a control sound in a space to which noise is transmitted from a noise source, a noise generation state detecting means for detecting a noise generation state of the noise source and outputting as a reference signal, Residual noise detecting means for detecting residual noise at a predetermined position and outputting the same as a residual noise signal, an adaptive digital filter having a variable filter coefficient, and filtering the reference signal with the adaptive digital filter to generate a drive signal of the control sound source. A drive signal generating means for generating, a transfer function filter that models a transfer function between the control sound source and the residual noise detecting means, and a result of filtering the reference signal with the transfer function filter and the residual noise signal. Adaptive processing means for updating the filter coefficient of the adaptive digital filter based on the noise in the space based on the noise. An active noise control device, comprising: a signal adding means capable of temporarily adding a pulse signal synchronized with the reference signal to the drive signal; and a filter coefficient variable and two identification functions for one transfer function. A filter, and setting the one filter for identification based on the residual noise signal when the signal adding means has performed the addition, and applying the residual noise signal when the signal adding means has not performed the addition. An identification filter setting means for setting the other identification filter based on the difference, and a transmission setting the transfer function filter based on a difference between a filter coefficient of the one identification filter and a filter coefficient of the other identification filter. An active noise control device comprising: a function filter setting unit. 2. A control sound source capable of generating a control sound in a space to which noise is transmitted from a noise source; a noise generation state detection means for detecting a noise generation state of the noise source and outputting the noise generation state as a reference signal; Residual noise detecting means for detecting residual noise at a predetermined position and outputting the same as a residual noise signal, an adaptive digital filter having a variable filter coefficient, and filtering the reference signal with the adaptive digital filter to generate a drive signal of the control sound source. A drive signal generating means for generating, a transfer function filter that models a transfer function between the control sound source and the residual noise detecting means, and a result of filtering the reference signal with the transfer function filter and the residual noise signal. Adaptive processing means for updating the filter coefficient of the adaptive digital filter based on the noise in the space based on the noise. An active noise control device, comprising: a signal adding means capable of temporarily adding a pulse signal synchronized with the reference signal to the drive signal; and a filter coefficient variable and two identification functions for one transfer function. A filter and a synchronized average value of the residual noise signal when the signal addition means performs the addition, as a filter coefficient of the one identification filter, and when the signal addition means does not perform the addition. Identification filter setting means for setting an average value synchronized with the residual noise signal as a filter coefficient of the other identification filter, and a filter coefficient of the one identification filter and a filter coefficient of the other identification filter And a transfer function filter setting means for setting the transfer function filter based on the difference between the two.