JP2003108151A - Noise reducing device - Google Patents

Abstract

PROBLEM TO BE SOLVED: To provide an active control type noise reducing device which can obtain excellent sound reduction effect over a wide frequency range. SOLUTION: An M-series signal sound is radiated by a dummy sound source 18 which is arranged nearby a noise source S. A control point microphone 25 is arranged at a control point C nearby the upper side of a soundproofing barrier 10 and a computer 26 computes a filter coefficient for reducing the M-series signal sound according to the output of the microphone and sets it. A filter performs a digital filtering process by using the digital value of the output of a sound source microphone 20 arranged nearby a dummy sound source 18 and the set filter coefficient and according to the filter output, a control speaker 14 radiates a control sound for reducing noise.

Description

Detailed Description of the Invention

[0001]

BACKGROUND OF THE INVENTION 1. Field of the Invention The present invention relates to a noise reduction device, and more particularly to an active control type noise reduction device that reduces noise by active control.

[0002]

2. Description of the Related Art According to the "Environmental White Paper" (2000 edition) of the Environmental Agency, noise is a problem that is deeply related to daily life among various pollutions, and that its sources are various. Therefore, the number of complaints accounts for the majority of pollution complaints. By source, noise from factories and business sites was the largest, followed by noise from construction work, which was a major cause of complaints. Conventionally, when noise from these factories and business sites and noise from construction work pose a problem, noise-reducing walls made of concrete blocks and reinforced concrete are used to reduce noise by adopting low-noise construction machinery. It was common to take measures such as installing noise barriers such as temporary enclosures for construction work made of iron plates of several millimeters to prevent noise. Further, in order to improve the noise reduction performance of the soundproof barrier, the conventional method had to provide a high soundproof barrier.

However, the provision of a high soundproof barrier entails problems such as increased material costs and labor costs, time-consuming work, and reinforcement of the barrier foundation due to an increase in the scale of the barrier. In addition, sound barriers are installed in consideration of harmony with the landscape of the surrounding area, so it may not be possible to secure the necessary barrier height to obtain the effect of noise prevention measures. For this reason,
When it is not possible to install a high soundproof barrier due to various reasons, measures have been taken such as unavoidably limiting the working time in order to improve the problem of noise.

In recent years, active noise control (ANC: Active Noise) has been proposed to reduce noise by active control.
Control) system is attracting attention. The silencing principle of ANC is "to superimpose a sound wave of opposite phase on an original sound wave to be muted". That is, as shown in FIG. 1, the control sound B radiated from the control sound source is superimposed on the noise A emitted from the noise source to reduce the sound pressure level.

For example, for a diffracted sound generated from a fixed noise source, diffracted at the apex of the soundproof barrier and transmitted to the outside of the barrier,
The above silencing principle can be applied. As shown in FIG. 2, the noise 12 generated from the noise source S is generated by the sound barrier 10
When passing through the vicinity of the apex (control point C) of D, the diffraction effect is exerted as a wave phenomenon. This means that the control point C becomes a new sound source (secondary sound source) centered on this point.
A control sound 16 is emitted from a control sound source (speaker) 14 installed near the sound barrier 10 to the control point C. At this time, the noise 12 from the noise source S at the control point C
The control sound 16 is processed so that the control sound 16 from the control speaker 14 has the same amplitude and opposite phase. This allows
An observation point O in the area of the sound barrier 10 opposite to the noise source S
By observing the noise at, the sound reduction effect of the sound barrier 10 or more can be obtained.

Therefore, by installing the ANC system in addition to the existing soundproof barrier, noise can be significantly reduced without a significant increase in cost due to the increase in height of the soundproof barrier. Therefore, noise barriers equipped with an ANC system generate noise from cooling towers and transformers, which are typical noise sources in factories and business sites, and construction machines such as bulldozers and power shovels, which are typical noise sources in construction work. Is expected as a technology to effectively prevent this.

[0007]

Conventionally, an ANC system provided in a soundproof barrier includes an adaptive digital filter.
(Adaptive filter) has been used. The adaptive digital filter is composed of hardware such as a DPS (digital signal processor), but typically, as shown in FIG. 2, a signal from a sound source microphone 20 installed near the noise source S FIR type and IIR type that perform digital filtering processing by calculating the product sum of the digital value (vector amount) Xk generated from the input digital value χk and the filter coefficient Wk (vector amount) when input Alternatively, a block 80 composed of a lattice type digital filter and LMS (Least Mean Square) which is a general filter coefficient updating algorithm
Using the algorithm, Newton's method, or steepest descent method, the residual signal E input from the control point microphone 25 is input.
It can be represented by k, that is, a block 82 that updates the filter coefficient so that the noise that remains without being muted is minimized, and a filter 84 that simulates the path from the control speaker 14 to the control point C.

However, the above-mentioned system using the adaptive filter has a problem that the predetermined frequency component having large power is muted and the other frequency components are hardly muted. Further, since not only the direct sound but also the delayed reflected sound from the surroundings are controlled, there is a problem that the volume reduction becomes small. Furthermore, since the entire frequency band to be controlled is controlled by one adaptive filter,
There is a problem that the load of the filter becomes large and control takes time.

The present invention has been made in view of the above circumstances, and a first object of the present invention is to provide an active control type noise reduction device capable of obtaining an excellent noise reduction effect over a wide frequency band. To do. A second object of the present invention is to provide an active control type noise reduction device with a large volume reduction. A third object of the present invention is to provide an active control type noise reduction device capable of high speed control.

[0010]

In order to achieve the first object, the noise reduction device according to the first invention is arranged close to a noise source so as to radiate an M-sequence signal sound. M
Based on the dummy sound source that emits the series signal sound, the first microphone arranged at the control point, the second microphone arranged near the dummy sound source, and the first microphone output, the M series signal sound is generated. A calculation unit that calculates a filter coefficient for reduction and a filter coefficient calculated by the calculation unit are set, and digital filtering processing is performed using the digital value of the second microphone output and the set filter coefficient. It is characterized by including a filter and a control sound emitting means for emitting a control sound for reducing a sound from a noise source based on the filter output.

In the noise reduction device according to the first aspect of the invention, the M-sequence signal sound having a wide frequency band is radiated from the dummy sound source arranged close to the noise source. A control point for controlling the noise emitted from the noise source is set, and the first microphone is arranged at this control point. The calculation means calculates a filter coefficient for reducing the M-sequence signal sound based on the output of the first microphone. The filter coefficient calculated by the calculating means is set in the filter. On the other hand, a second microphone is arranged near the dummy sound source. The filter performs digital filtering processing using the digital value of the second microphone output and the set filter coefficient. Then, the control sound emitting means emits a control sound for reducing the sound from the noise source based on the filter output.

In the present invention, since the filter coefficient is calculated so as to reduce the M-sequence signal sound, a filter in which this filter coefficient is set is used and the control sound is radiated based on the filter output. An excellent sound reduction effect can be obtained over the same wide frequency band as the M-sequence signal sound.

In order to achieve the above-mentioned second object, the noise reducing device according to the second invention radiates the first microphone arranged at the control point and the vicinity of the noise source or an M-sequence signal sound. The second microphone arranged near the dummy sound source arranged close to the noise source and the filter point for reducing the sound from the noise source based on the output of the first microphone. And a filter coefficient calculated by the calculating means is set, and the digital value of the second microphone output and the set filter coefficient are used. A filter for digital filtering processing, and a control sound emitting means for emitting a control sound for reducing a sound from a noise source based on the output of the filter are configured to be included.

In the noise reducing device according to the second aspect of the present invention, a control point for controlling the noise radiated from the noise source is set, and the first microphone is arranged at this control point. The calculation means calculates the filter coefficient for reducing the sound from the noise source based on the output of the first microphone so that only the direct sound to the control point is reduced. The filter coefficient calculated by the calculating means is set in the filter. on the other hand,
A second microphone is arranged near the noise source. The filter performs digital filtering processing using the digital value of the second microphone output and the set filter coefficient. Then, the control sound emitting means emits the control sound for reducing the direct sound from the noise source based on the filter output.

In the case of controlling the delayed reflected sound from the surroundings as well, the volume reduction is reduced, but the filter coefficient is calculated so that only the direct sound to the control point is reduced. The sound component can be effectively muted, and the direct reflected sound component can be effectively muted, so that the subsequent reflected sound component can also be muted. As a result, a large volume reduction can be obtained as a whole.

In the noise reduction device according to the first and second aspects of the present invention, it is possible to provide a computing means for computing the filter coefficient by an explicit method as the computing means. By calculating the filter coefficient by the explicit method, the impulse response measured in the calculation process can be windowed and only the direct sound can be reduced. Further, the calculation means of the noise reduction device according to the first aspect may be configured to calculate the filter coefficient so that only the direct sound to the control point is reduced.

In order to achieve the third object, the noise reducing device according to the third invention comprises a first microphone arranged at a control point and a second microphone arranged near a noise source. , A dividing means for dividing the second microphone output into sounds of a plurality of bands, and a digital value and a filter coefficient of the sound of each band which is provided corresponding to each of the plurality of bands and is divided by the dividing means. A plurality of filters for digital filtering using the changing means, changing means for changing the filter coefficient of each of the filters so as to minimize the deviation between the filter output and the target value, and a sound from a noise source based on the filter output. And a control sound emitting means for emitting a control sound for reducing the noise.

In the noise reduction device according to the third aspect of the present invention, a control point for controlling the noise radiated from the noise source is set, and the first microphone is arranged at this control point. on the other hand,
A second microphone is arranged near the noise source. The output of the second microphone is divided into sounds of a plurality of bands by the dividing means, and is input to a plurality of filters provided corresponding to each of the plurality of bands. Multiple filters
Digital filtering processing is performed using the digital value of the sound in each band divided by the dividing means and the filter coefficient. The filter coefficient of each of the plurality of filters is changed by the changing means so that the deviation between the filter output and the target value is minimized. Then, the control sound emitting means emits a control sound for reducing the sound from the noise source based on the filter output.

In this way, the noise is divided into sounds of a plurality of bands,
Since the processing is performed by the plurality of filters provided corresponding to each of the plurality of bands, the tap length of the filter becomes short. This enables high speed control. When the noise is divided into the frequency band component in which noise is dominant at the control point and the other frequency band components, the load on each filter is further reduced and excellent noise reduction effect is obtained in all frequency bands. be able to.

In the noise reduction device according to the above-mentioned first to third inventions, the control point is set in the vicinity of the upper side of the wall surface arranged so as to surround the noise source, and the control sound radiating means is provided from the noise source. It can be configured to reduce the sound emitted and diffracted at the upper edge of the wall. Therefore, even if a high soundproof barrier cannot be installed due to various reasons, the noise can be significantly reduced by installing the noise reduction device on the existing soundproof barrier. Further, it is preferable to use a control sound emitting means in which a plurality of sound sources are arranged in a line or a plane so that the control sound is emitted in the same direction.

[0021]

BEST MODE FOR CARRYING OUT THE INVENTION Embodiments of the present invention will be described below with reference to the drawings. (First Embodiment) An embodiment in which the noise reduction device of the present invention is applied to a soundproof system at a construction site will be described. As shown in FIG. 3, the soundproof system according to the present embodiment includes a soundproof barrier 10 and a soundproof barrier arranged around the noise source S to reduce noise generated from the noise source S such as a power shovel. A control speaker 14 that emits a control sound 16 toward a control point C set near the upper side of the wall surface of 10 and a noise source S that is arranged so as to emit an M-sequence signal sound having a flat frequency characteristic. Dummy sound source 1
8. Sound source microphone 2 arranged near the noise source S
0, a digital filter device 24 having a digital filter 22, a control point microphone 25 arranged near the control point C, and a computer 26.

The soundproof barrier 10 is formed by vertically leaning a plurality of metal plates such as iron plates against the ground, and each of the metal plates is fixed at a predetermined position by a mounting member or the like. Further, as shown in FIG. 4 (A), the sound insulation enhancement panel 1
It is preferable to arrange 1 on the noise source S side of the soundproof barrier 10 in parallel with the soundproof barrier 10 and at a predetermined distance from the soundproof barrier 10. It is preferable that the sound insulation enhancing panel 11 has a sound insulation performance of 10 dB or more in the low sound range, as shown in FIG.
As shown in (A), it is more preferable that the sound absorbing material 13 such as glass wool is attached to the panel surface on the noise source S side. Furthermore, the soundproof barrier 10 and the sound insulation enhancement panel 11
Is more preferably weather resistant because it is installed outdoors and used.

The control speaker 14 is, as shown in FIG.
The support member 4 is attached to the tip of a rod-shaped support member 48 that is configured to extend and contract in the vertical direction (arrow A direction), and
The tip of 8 is an axis, and is rotatable around this axis. The support 48 is fixed to an installation base 50 that is movable in the horizontal direction (the direction of arrow B), such as having wheels at the bottom. It is preferable that the installation base 50 be movable by several workers who work on site. With these configurations, the height at which the control speaker 14 is arranged, the distance from the soundproof barrier 10, and the output direction of the speaker can be freely changed. It should be noted that the control speaker 14 is arranged at a distance exceeding one wavelength of the noise wave from the upper side of the soundproof barrier 10 so that the control sound is diffracted by the soundproof barrier 10.

When the noise source S moves such as a power shovel, the dummy sound source 18 is connected to the receiver 32 arranged near the noise source S via the amplifier 34. Then, the dummy sound source 18 is mounted on the moving body together with the receiver 32 and the amplifier 34. On the other hand, the computer 26
The M-series signal generator 28 that generates the M-series signal based on the instruction from is placed at a location away from the noise source S and is connected to the transmitter 30. The signal generated by the M-sequence signal generator 28 is transmitted from the transmitter 30 to the receiver 3
2 is wirelessly transmitted, amplified by the amplifier 34, and input to the dummy sound source 18. Then, the M-sequence signal sound is radiated from the dummy sound source 18.

The dummy sound source 18 can be constituted by a directional speaker that changes the directivity of the speaker to a directivity approximately equivalent to the directivity of the noise source S by the built-in DPS processing system. As a result, the sound wave reaching the control point C from the dummy sound source 18 follows substantially the same route as the sound wave reaching the control point C from the noise source S. Further, when the dummy sound source 18 is mounted on a moving body such as a power shovel, it is arranged in the vicinity of a noise source S such as an exhaust pipe of the power shovel or an engine, so that it is small and easy to install. Those are preferable.

When the noise source S moves like a power shovel or the like, the sound source microphone 20 uses the preamplifier 3
6 to the transmitter 38. Then, the sound source microphone 20 is mounted on the moving body together with the preamplifier 36 and the transmitter 38. On the other hand, the digital filter device 2
4 is arranged at a place away from the noise source S and is connected to the receiver 40. The sound wave picked up by the sound source microphone 20 is amplified by the amplifier 36, wirelessly transmitted from the transmitter 38 to the receiver 40, A / D (analog / digital) converted by an A / D converter (not shown), and a digital signal. Is input to the digital filter device 24.

The digital filter device 24 comprises a digital filter 22 and an amplifier 42. The digital filter 22 is connected to the control speaker 14 and the computer 26 via the amplifier 42. A howling suppression circuit that suppresses howling may be inserted between the digital filter 22 and the amplifier 42. The filter coefficient of the digital filter 22 is calculated by the computer 26 and set in the digital filter 22. The method of calculating the filter coefficient will be described later. The digital filter 22 performs a digital filtering process using the digital signal input from the sound source microphone 20 and the set filter coefficient. The filtered signal is D / A converted by a D / A converter (not shown), amplified by the amplifier 42, and output to the control speaker 14. Then, a sound wave corresponding to the filtered signal is emitted from the control speaker 14 as a control sound 16.

The control point microphone 25 is connected to the computer 26 via an amplifier 44. The control point microphone 25 collects the sound wave emitted from the control speaker 14 together with the M-sequence signal sound emitted from the dummy sound source 18 or the noise emitted from the noise source S. The collected sound waves are amplified by the amplifier 44, A / D converted by an A / D converter (not shown), and the computer 26
Entered in.

The computer 26 is provided with a central processing circuit (CPU), a ROM and a RAM.
Then, the filter coefficient of the digital filter 22 is calculated, and the calculated filter coefficient is set in the digital filter 22. Further, the computer 26 is connected to a display device 46 such as a display, and the control point microphone 2 is connected.
The noise level at the control point C can be appropriately displayed on the display device 46 based on the input signal from the control unit 5. Thereby, the noise level at the control point C can be constantly monitored, and the effect of the control sound emission can be confirmed.

Next, the operation of the soundproof system according to the present embodiment will be described separately for the filter coefficient setting operation and the noise control operation.

First, the operation of setting the filter coefficient in the digital filter 22 will be described. The arrangement position and the output direction of the control speaker 14 and the dummy sound source 18 are adjusted so that the sound wave is emitted toward the control point C near the upper side of the wall surface of the soundproof barrier 10. After this adjustment is completed, the dummy sound source 1
8 is operated to set the filter coefficient in the digital filter 22. It is also possible to set a plurality of control points C. When a plurality of control points C are set, the control speaker 14 is arranged for each control point.

The M-sequence signal generator 28 is the computer 2
An M-sequence signal is generated according to the instruction from 6. The signal generated by the M-sequence signal generator 28 is input to the dummy sound source 18 via the transmitter 30, the receiver 32, and the amplifier 34. Then, the dummy sound source 18 emits the M-sequence signal sound toward the control point C. The emitted M-sequence signal sound is
The sound source microphone 20 collects sound, and the digital filter device 2 is passed through the amplifier 36, the transmitter 38, and the receiver 40.
4 is input.

The input digital signal is delayed by the digital filter 22 and output to the control speaker 14 via the amplifier 42. And M corresponding to the digital signal
A series signal sound is emitted from the control speaker 14 toward the control point C. By the control point microphone 25, the M-sequence signal sound radiated from the dummy sound source 18 and the control speaker 14
The M-sequence signal sound radiated from is collected. The collected sound wave is input to the computer 26 via the amplifier 44.

Here, the transfer function of the digital filter 22 is assumed to be W (ω), and the coefficient of the impulse response of the transfer function W (ω) is set to δ (t-t _delay ). t _delay is a delay time (ms) and can be set to 300 ms, for example.

A part of the M-sequence signal sound radiated from the dummy sound source 18 directly reaches the control point microphone 25 installed above the soundproof barrier 10. The transfer function of the transfer path A is A (ω). Further, a part of the radiated M-sequence signal sound is picked up by the sound source microphone 20 installed near the dummy sound source 18, passes through the digital filter device 24 (delayed by the digital filter 22), and is output from the control speaker 14. It is emitted and reaches the control point microphone 25. Let the transfer function of this transfer path B be B (ω).
The control point microphone 25 picks up the signal sound from the path A and the signal sound from the path B at the same time.

Dummy sound source 18 to sound source microphone 20
If the transfer function of the path C1 from the control speaker 14 to the control point microphone 25 is C2 (ω), the transfer function of the path B is B1.
(Ω) is represented by the following formula. Note that C1 (ω) and C2
Let the product of (ω) be the transfer function C (ω) of the route C (route C1 + C2).

B (ω) = C1 (ω) C2 (ω) W (ω)
= C (ω) W (ω) where W (ω) = 1, the transfer function B of the path B
(Ω) becomes equal to the transfer function C (ω) of the path C.

The computer 26 calculates the filter coefficient of the digital filter 22 by the explicit method according to the following procedure. The explicit method here is to measure the impulse response of the path A and the impulse response of the path B in advance,
This is a method of calculating the filter coefficient by numerical calculation.

The digital signal taken into the computer 26 from the control point microphone 25 is converted into an impulse response by Hadamard conversion. As shown in FIG. 5A, the impulse response of the path C is delayed by t _delay (ms), and the impulse responses of the path A and the path C are temporally separated from each other. Windows 70, 7
The impulse response can be cut out for each path by multiplying by 2. Let the impulse response of path A be the function a (t)
And the impulse response of path C is represented by the function c (t). Further, as shown in FIG. 5B, by multiplying the impulse response a (t) of the path A by the time window 52, only the impulse response of the direct sound can be cut out.

The c (t−t _delay ) obtained by rewinding c (t) by the delay time is set again as c (t). In addition, the impulse response of the path A, which is obtained by cutting out only the portion corresponding to the direct sound, is a-t again. And route A
Fourier transform of the impulse response a (t) of A to obtain a transfer function A (ω), and Fourier transform of the impulse response c (t) of the path C to obtain a transfer function C (ω).

Assuming that the transfer function A (ω) of the path A and the transfer function B (ω) of the path B satisfy the silencing condition of A (ω) + B (ω) = 0, the transfer of the digital filter 22. The function W (ω) is expressed by the following equation. W (ω) =-A (ω) / C (ω)

Then, W (ω) is calculated using the transfer function A (ω) and the transfer function C (ω) obtained by the Fourier transform, and this is inverse Fourier transformed to obtain the function w (t). . The coefficient of this function w (t) is the digital filter 2
2 is a filter coefficient. The function w (t) may be directly obtained in the time domain by matrix calculation from the relationship shown in the following equation. w (t) * c (t) =-a (t) (* represents a convolution operation.)

The computer 26 inputs the obtained filter coefficient to the digital filter device 24 and sets it in the digital filter 22. As a result, the digital filter 22 becomes a filter having a transfer function of W (ω). Digital filter 22 in which the above filter coefficient is set
Is an inverse filter that generates a control sound capable of canceling noise in the same wide frequency band as the M-sequence signal sound.

Next, the noise control operation will be described.

The noise 12 radiated from the noise source S is picked up by the sound source microphone 20, and is amplified by the amplifier 36 and the transmitter 3.
8 and the receiver 40, and is input to the digital filter device 24 as a digital signal. On the other hand, the digital filter 22 performs a digital filtering process using a digital signal input from the sound source microphone 20 and a set filter coefficient according to an instruction from the computer 26. The filtered signal is output to the control speaker 14 via the amplifier 42. Then, the sound wave corresponding to the filtered signal is
A control sound 16 is emitted from the control speaker 14 toward the control point C, and at the control point C, the noise 12 from the noise source S is emitted.
However, it is canceled by the control sound 16.

By the control point microphone 25, the noise source S
The noise 12 radiated from the control speaker 16 and the control sound 16 radiated from the control speaker 14 are collected. In reality, the noise 12 and the control sound 16 cancel each other, so that only a small sound wave is collected. When the collected sound waves are input to the computer 26 via the amplifier 44, the computer 26
Displays the noise level at the control point C on the display device 46 based on this input signal.

As described above, in the present embodiment, the M-sequence signal tone having a flat frequency characteristic is radiated from the dummy sound source arranged near the noise source to identify the inverse filter. Therefore, the obtained inverse filter is obtained. Can be used to cancel noise in a wide frequency band. As shown in FIG. 6, when the inverse filter is identified by the sound wave generated from the actual noise source, the low-frequency 50 Hz component of high power is silenced, and other frequency components are hardly attenuated. It can be seen that, when the M-sequence signal sound is emitted and the inverse filter is identified, the noise is canceled in a wide frequency band. For example, even if the frequency spectrum changes due to changes in the engine speed of power shovels,
Correspondingly, noise can be canceled.

Further, in the conventional filtered-X method and the LMS method, in the identification of the inverse filter, not only the direct sound but also the delayed reflected sound from the surroundings are controlled, so that the volume reduction becomes small. However, in the present embodiment,
Since the filter coefficient is calculated by the explicit method, as shown in FIG. 5, the impulse response measured in the calculation process can be windowed, and only the direct sound component can be muted selectively. By effectively muting the direct sound component, the subsequent reflected sound component can also be damped, and a large sound damping effect can be obtained as a whole.

Note that the filter coefficient may be calculated using the cross spectrum method described later, instead of the method of calculating the filter coefficient of the present embodiment.

(Second Embodiment) As shown in FIG. 7, the soundproof system according to the present embodiment is different from the first embodiment except that the filter coefficient is set in the digital filter without using the dummy sound source. Since the configuration is the same as that of the embodiment, the same reference numerals are given to the same portions and the description thereof will be omitted.

When the noise source S moves, such as in a power shovel, the sound source microphone 20 uses the preamplifier 3
6 to the transmitter 38. Then, the sound source microphone 20 is mounted on the moving body together with the preamplifier 36 and the transmitter 38. On the other hand, the digital filter device 2
The computer 4 and the computer 26 are arranged at a place apart from the noise source S and are connected to the receiver 40, respectively. The sound waves picked up by the sound source microphone 20 are amplified by the amplifier 36, wirelessly transmitted from the transmitter 38 to the receiver 40,
The digital signal is A / D converted by an A / D converter (not shown) and input to the digital filter device 24 and also to the computer 26.

Next, the operation of the soundproof system according to this embodiment will be described. Since the noise control operation is similar to that of the first embodiment, the description thereof is omitted, and only the operation of setting the filter coefficient to the digital filter 22 will be described.

The position and the output direction of the control speaker 14 are adjusted so that the sound wave is emitted toward the control point C near the upper side of the wall surface of the soundproof barrier 10. After this adjustment is completed
The filter coefficient is set in the digital filter 22.

The noise radiated from the noise source S is picked up by the sound source microphone 20 and inputted to the digital filter device 24 and the computer 26 as a digital signal via the amplifier 36, the transmitter 38 and the receiver 40. . The input digital signal is delayed by the digital filter 22 and output to the control speaker 14 via the amplifier 42. Then, noise corresponding to the digital signal is radiated from the control speaker 14 toward the control point C. The control point microphone 25 picks up the noise emitted from the noise source S and the noise emitted from the control speaker 14. The collected sound waves are input to the computer 26 via the amplifier 44.

Here, the transfer function of the digital filter 22 is assumed to be W (ω), and the coefficient of the impulse response of the transfer function W (ω) is set to δ (t−t _delay ). t _delay is a delay time (ms) and can be set to 300 ms, for example.

A part of the noise radiated from the noise source S reaches the control point microphone 25 installed above the sound barrier 10 directly. This transmission path is called a path A. The transfer function of the path A is A (ω). A part of the radiated noise is picked up by the sound source microphone 20 installed near the noise source, reaches the control point microphone 25 via the digital filter device 24 (delayed by the digital filter 22). This transmission path is called a path B. The transfer function of the path B is B (ω). Control point microphone 2
In 5, the signal sound from the path A and the signal sound from the path B are picked up at the same time.

The transfer function of the path C1 from the noise source S to the sound source microphone 20 is C1 (ω), and the control speaker 1
Assuming that the transfer function of the path C2 from 4 to the control point microphone 25 is C2 (ω), the transfer function of the path B is B (ω).
Is represented by the following formula. The product of C1 (ω) and C2 (ω) is the transfer function C (ω) of the route C (route C1 + C2).
And B (ω) = C1 (ω) C2 (ω) W (ω) = C (ω) W
(Ω)

Here, when W (ω) = 1, the transfer function of the path B is equal to B (ω) and the transfer function C (ω) of the path C.

The computer 26 calculates the filter coefficient of the digital filter 22 by the explicit method using the cross spectrum method in the following procedure. The explicit method here is a method in which the impulse response of the route A and the impulse response of the route B are measured in advance, and the filter coefficient is calculated by numerical calculation.

A digital signal taken from the sound source microphone 20 to the computer 26 is cut out in a predetermined time window, and the cut out signal is represented by a function x (t). On the other hand, the digital signal taken into the computer 26 from the control point microphone 25 is cut out in a predetermined time window, and the cut out signal is represented by a function y (t).

Autocorrelation function Φ of the function x (t)
_xx (t), a cross-correlation function Φ _xy (t) between the function x (t) and the function y (t) is obtained. S / N is improved by performing synchronous addition on Φ _xx (t) and Φ _xy (t). Based on the result, the autocorrelation function Φ _xx (t) and the cross-correlation function Φ _{xy are} re-established.
Represents (t). Fourier transform of both functions Φ _xx
(Ω) and the function Φ _xy (ω). Note that the function Φ
_xx (ω) and the function Φ _xy (ω) may be obtained by synchronously adding those obtained in the frequency domain.

Calculate Z (ω) = Φ _xy (ω) / Φ _xx (ω). Inverse Fourier transform of Z (ω) is the function z
(T). The function z (t) includes the impulse responses of the paths A and C, but the impulse response of the path C is delayed by t _delay (ms), and the impulse responses of the paths A and C are Since they are separated in time, it is possible to multiply them by time windows to cut out the impulse response for each path. The impulse response of the path A is represented by the function a (t), and the impulse response of the path C is represented by the function c (t). Furthermore, the impulse response a of path A
By multiplying (t) by a time window, only the impulse response of the direct sound can be cut out.

The c (t−t _delay ) obtained by rewinding c (t) by the delay time will be referred to as c (t) again. In addition, the impulse response of the path A, which is obtained by cutting out only the portion corresponding to the direct sound, is a-t again. And route A
Fourier transform of the impulse response a (t) of A to obtain a transfer function A (ω), and Fourier transform of the impulse response c (t) of the path C to obtain a transfer function C (ω).

Assuming that the transfer function A (ω) of the path A and the transfer function B (ω) of the path B satisfy the silencing condition of A (ω) + B (ω) = 0, the transfer of the digital filter 22. The function W (ω) is expressed by the following equation. W (ω) =-A (ω) / C (ω)

Transfer function A obtained by Fourier transform
W (ω) is calculated using (ω) and the transfer function C (ω), and this is inverse Fourier transformed to obtain the function w (t).
The coefficient of this function w (t) is the filter coefficient of the digital filter 22.

The computer 26 inputs the obtained filter coefficient to the digital filter device 24 and sets it in the digital filter 22. As a result, the digital filter 22 becomes a filter having a transfer function of W (ω). Digital filter 22 in which the above filter coefficient is set
Is an inverse filter that produces a control sound that can cancel noise.

As described above, in the present embodiment, since the filter coefficient is calculated by the explicit method using the cross spectrum method, the impulse response measured in the calculation process can be windowed, and only the direct sound component is obtained. You can mute selectively. By effectively muting the direct sound component, the subsequent reflected sound component can also be damped, and a large sound damping effect can be obtained as a whole.

The M-sequence signal can be converted into an impulse response by Hadamard transform, but since the M-sequence signal is not generated from the actual noise source, the impulse response cannot be obtained by Hadamard transform. According to the cross spectrum method, since the impulse response can be obtained even when the M-sequence signal is not used, the filter coefficient can be calculated by the explicit method without using the dummy sound source, and the device configuration becomes simple.

(Third Embodiment) As shown in FIG. 8, the soundproof system according to the present embodiment uses a frequency band division system without using a dummy sound source and a digital filter device.
Since the configuration is the same as that of the second embodiment except that the adaptive filter is identified using the Filtered-X method,
The same parts are designated by the same reference numerals and the description thereof will be omitted.

The receiver 40 that receives the signal from the sound source microphone 20 is connected to a high-pass filter 54 that transmits only high-frequency components and a low-pass filter 56 that transmits only low-frequency components. The high pass filter 54 is
Connected to the adaptive digital filter 58 on the high frequency side,
The low-pass filter 56 is connected to the low frequency side adaptive digital filter 60. Control point microphone 2
5 is also connected via an amplifier 44 to a high-pass filter 62 that transmits only high-frequency components and a low-pass filter 64 that transmits only low-frequency components. The high pass filter 62 is connected to the high frequency side adaptive digital filter 58, and the low pass filter 64 is connected to the low frequency side adaptive digital filter 60. Note that the adaptive digital filters 58 and 60 have the same configurations as the conventional ones, and thus the description thereof will be omitted.

Of the received signal from the sound source microphone 20, only the high frequency component is input to the high frequency side adaptive digital filter 58, and only the low frequency component is input to the low frequency side adaptive digital filter 60. Further, only the high frequency component of the residual signal from the control point microphone 25 is input to the high frequency side adaptive digital filter 58,
Only the low frequency component is input to the low frequency side adaptive digital filter 60. In each of the adaptive digital filters 58 and 60, the filter coefficient is updated so that the residual signal component is minimized. Then, in each of the adaptive digital filters 58 and 60, the input signal is divided into a low frequency component and a high frequency component for filtering, and the digital signal is output to the control speaker 14 via the amplifier 66.

As described above, in the present embodiment, the frequency band is divided into the high frequency band and the low frequency band to produce the inverse filter. Therefore, as shown in FIG. 9, all frequencies to be controlled are controlled. Compared with the case where the band is controlled by one inverse filter, the tap length is shortened and high speed control becomes possible.

In particular, when the inverse filter is identified by the sound wave generated from the noise source, as described above, the low-frequency 50 Hz component of high power is muted and the other frequency components are hardly attenuated. At each point, the load of each inverse filter is reduced by dividing the component of the frequency band (for example, low frequency range of 80 Hz or less) band and the components of the other frequency bands into the inverse filters to identify them. As shown in FIG. 13, the sound reduction effect can be obtained in the entire frequency band, and high-speed control can be performed.

In the above first to third embodiments,
The control speaker may be configured by a point sound source, but may also be configured by a linear array sound source in which a plurality of sound sources are linearly arranged so that the control sound is emitted in the same direction. Even if the soundproof barrier 10 is arranged at the same position with respect to the noise source S that radiates the noise 12, as shown in FIG. 11, when the control speaker 14 is composed of a point sound source, Since the radiated sound propagates in almost all directions and the sound waves propagate in areas other than the control target area, there is a region where noise increases on the outside of the soundproof barrier 10. However, as shown in FIG. When 14 is composed of a linear array sound source, the directivity of the control speaker 14 becomes extremely sharp and the sound wave propagates only to the control target area, so that the noise is reduced in the entire area outside the sound barrier 10. . It is also possible to suppress howling that occurs when sound waves emitted from the control speaker are picked up by the sound source microphone.

Further, the control speaker may be composed of a planar array sound source in which a plurality of sound sources are arrayed in a plane so that the control sound is emitted in the same direction. By using the planar array sound source as the control speaker, noise can be reduced in the entire area outside the sound barrier, and howling can be significantly suppressed.

In the first and second embodiments described above, when the distance from the noise source S to the control point C is long,
It takes time for the sound wave radiated from the noise source S to reach the control point microphone (for example, the distance from the noise source S to the control point C is 100 ms at 34 m), and the sound source directly reaches the control point microphone. Since it becomes difficult for a computer to simultaneously process the signal and the signal passing through the digital filter device, it is preferable to insert a delay circuit between the digital filter device and the control speaker.

In the third embodiment described above, it is preferable that the sampling frequency of the low-pass filter is lower than the sampling frequency of the high-pass filter.
The low frequency component of the sound wave may not be converged in frequency and the impulse response length may be long, but by reducing the sampling frequency of the low pass filter, the tap length of the adaptive digital filter on the low frequency side can be increased. , The filter coefficient can be calculated.

For example, as shown in FIG. 12, a downsampling circuit 74 is inserted between the lowpass filter 64 and the adaptive digital filter 60, and a downsampling circuit 75 is inserted between the lowpass filter 56 and the adaptive digital filter 60. And the upsampling circuit 76 is inserted between the adaptive digital filter 60 and the amplifier 66, the sampling frequency of the low-pass filter can be reduced. Also,
As shown in FIG. 12, when the distance from the noise source S to the control point C is long, the high-frequency adaptive digital filter 58 and the low-frequency side are provided for the same reason as in the first and second embodiments. Delay circuit 7 in the adaptive digital filter 60
It is preferable to insert 7, 78, respectively.

[0079]

The noise reduction device according to the first aspect of the present invention can obtain an excellent noise reduction effect over a wide frequency band.
Has the effect. The noise reduction device according to the second aspect of the invention has an effect that a large volume reduction can be obtained. The noise reduction device according to the third aspect of the invention has an effect that high-speed control is possible.

[Brief description of drawings]

[Figure 1] Active noise control (ANC)
It is a figure for demonstrating the silencing principle of.

FIG. 2 is a schematic view showing a configuration of a soundproof barrier provided with a conventional ANC system.

FIG. 3 is a schematic diagram showing a configuration of a soundproof system according to the first embodiment.

FIG. 4 is a schematic view showing a support mechanism of a control speaker.

FIG. 5 is a diagram showing an impulse response based on an input of a control point microphone.

FIG. 6 is a graph showing the frequency dependence of the noise level before and after muffling by ANC in comparison between the soundproof system according to the first embodiment and the conventional soundproof system.

FIG. 7 is a schematic diagram showing a configuration of a soundproofing system according to a second embodiment.

FIG. 8 is a schematic diagram showing a configuration of a soundproofing system according to a third embodiment.

FIG. 9 is a soundproof system according to a third embodiment of the ANC.
It is a graph which shows the frequency dependence of the noise level at the time of silencing by.

FIG. 10 is a plan view showing an example in which a control speaker composed of a point sound source is arranged.

FIG. 11 is a plan view showing an example in which a control speaker configured by a linear array sound source is arranged.

FIG. 12 is a schematic view showing a modified example of the soundproofing system according to the third embodiment.

FIG. 13 is a graph showing the frequency dependence of the noise level before and after performing the band division in comparison between the soundproof system according to the third embodiment and the conventional soundproof system.

[Explanation of symbols]

S noise source C control point 10 Sound barrier 14 Control speaker 18 Dummy sound source 20 sound source microphone 22 Digital filter 24 Digital filter device 25 control point microphone 26 Computer 28 M series signal generator

   ─────────────────────────────────────────────────── ─── Continued front page    (72) Inventor Yasushi Nakayama             Chiba Prefecture Inzai City 1-5 Otsuka 1 Stock Association             Takenaka Corporation Technical Research Institute F-term (reference) 5D061 FF02

Claims (7)

[Claims] 1. A dummy sound source arranged near a noise source to emit M-sequence signal sound, a first microphone arranged at a control point, and a second microphone arranged near the dummy sound source. , A calculating means for calculating a filter coefficient for reducing the M-sequence signal sound based on the first microphone output, the filter coefficient calculated by the calculating means is set, and a digital signal of the second microphone output is set. A noise reduction that includes a filter that performs digital filtering processing using a value and a set filter coefficient, and a control sound emitting unit that emits a control sound for reducing a sound from a noise source based on the filter output. apparatus. 2. The noise reduction device according to claim 1, wherein the calculation means calculates the filter coefficient so that only the direct sound to the control point is reduced. 3. A first microphone arranged at a control point and a noise source, or a dummy sound source arranged near the noise source so as to emit an M-sequence signal sound. A second microphone, an arithmetic means for operating the filter coefficient for reducing the sound from the noise source based on the output of the first microphone so that only the direct sound to the control point is reduced, and the arithmetic means The filter coefficient calculated by the above is set, and a filter that performs digital filtering processing using the digital value of the second microphone output and the set filter coefficient, and the sound from the noise source is reduced based on the filter output. A noise reducing device including: a control sound emitting means for emitting a control sound for performing the operation. 4. The noise reduction device according to claim 2, wherein the calculation means calculates the filter coefficient by an explicit method. 5. A first microphone arranged at a control point, a second microphone arranged in the vicinity of a noise source, a dividing means for dividing the output of the second microphone into sounds of a plurality of bands, and A plurality of filters that are provided corresponding to each of the plurality of bands and that perform digital filtering processing using the digital value of the sound in each band divided by the dividing means and the filter coefficient, and the deviation between the filter output and the target value. Noise reduction means for changing the filter coefficient of each of the filters so that the noise is minimized, and a control sound emitting means for emitting a control sound for reducing the sound from the noise source based on the filter output. apparatus. 6. The control point is set in the vicinity of an upper side of a wall surface arranged so as to surround the noise source, and the control sound emitting means reduces a sound emitted from the noise source and diffracted by the upper side of the wall surface. The noise reduction device according to claim 1. 7. The noise reduction according to claim 1, wherein the control sound emitting means has a plurality of sound sources arranged linearly or in a plane so that the control sound is emitted in the same direction. apparatus.