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DESCRIPTION JP2002055684

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DESCRIPTION JP2002055684
[0001]
BACKGROUND OF THE INVENTION 1. Field of the Invention The present invention generates a
secondary sound having the same amplitude and opposite phase as noise, and causes this to
interfere with the noise to cancel it, and feeds back the signal after the interference to generate a
secondary sound. The present invention relates to a feedback-type active noise control device
used to generate
[0002]
2. Description of the Related Art As a technique for reducing noise, a technique of generating a
secondary sound having the same amplitude and opposite phase as noise and making it interfere
with noise, thereby canceling noise with secondary sound and reducing noise Is being studied. As
an application example of such a technique, for example, a case applied to an ear protector
shown in JP-A-63-503186, a case applied to an audio headphone shown in JP-A-6-343195, etc.
There is.
[0003]
FIG. 9 is a schematic view showing an example of a headphone using a conventional feedback
type active noise control device. In the figure, 11 is a headphone, 12 is a diaphragm, and 13 is a
microphone. In the example shown in FIG. 9, a local sealed space is formed between the
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diaphragm 12 of the headphone 11 (or ear protector) and the ear canal, and the small
microphone 13 is disposed in the closed space in proximity to the diaphragm 12. There is. An
electrical processing C is added to the signal of the microphone 13 to form a so-called feedback
circuit that combines with the desired electrical signal to be reproduced, thereby obtaining an
output acoustic signal with significantly reduced external noise and distortion. be able to.
[0004]
In such a configuration, a disturbance input such as noise is collected by the microphone 13, and
a signal having the same amplitude and the opposite phase as the disturbance input is
superimposed on the input signal and emitted from the diaphragm 12 as a secondary sound. As a
result, the disturbance input and the sound emitted from the diaphragm 12 interfere with each
other to cancel the disturbance input. Therefore, the listener can hear only the sound of the input
signal.
[0005]
FIG. 10 is a block diagram showing an example of a general feedback system. In general, the
feedback system is known to have an effect of suppressing the influence of the disturbance input
d on the output y and an effect of suppressing the influence of the characteristic variation of the
system on the output y. These effects are obtained by multiplying P (s), C (s) and H (s) in FIG. 10,
so-called cycle transfer characteristic G0 (s) = P (s) × C (s) in this feedback system. It is known
that the higher the gain characteristic of H (s), the higher it becomes. However, if the gain
characteristic of the open loop transfer characteristic G0 (s) is increased indiscriminately, the
stability of the entire system is impaired and the risk of oscillation increases, so the disturbance
suppression effect, parameter variation suppression effect, and stability It is necessary to
determine the characteristics of the compensation elements C (s) and H (s) while taking tradeoffs.
[0006]
In the case of forming a feedback circuit as an active noise control device, the time delay
required for the sound wave to reach the error detection means such as the microphone from the
speaker emitting the secondary sound becomes a major issue. The time delay element is
generally represented by the time delay element = Exp (-jτω) (1) in the frequency domain. Here,
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τ represents a delay time, and ω represents an angular frequency. The equation (1) represents
a characteristic of rotating on a circle having a size of a radius of 1 on a real-imaginary plane.
This is considered from the Nyquist stability discriminant which specifies the stability of the
feedback system, and at a frequency at which the time delay element passes (−1, 0) on the realimaging plane, P (s) × C ( s) It means that the gain of × H (s) can not be made 1.0 or more. That
is, the larger the time delay included in the system, the more the loop transfer characteristic
passes (-1, 0) on the real-imaging plane within a certain frequency range, and the design of the
compensator becomes more difficult. Become. This is the reason why the feedback type active
noise control device is limited to local noise control, that is, noise control in a space where the
time delay is very short.
[0007]
In a very narrow enclosed space such as the headphone shown in FIG. 9 described above, the
distance from the diaphragm 12 to the microphone 13 can be made extremely short, and the
time delay can be made very short. it can. However, application to applications other than such
applications has been difficult.
[0008]
In recent years, attempts have been made to achieve both stability against time delay and control
performance. For example, Journal of the Japan Society of Mechanical Engineers (C edition), Vol.
62, no. 597, Nomin et al., "Active Noise Control of One-Dimensional Exhaust Duct System
Applying H.INF. As shown in 157-162, there is a design case to which H 制 御 control theory is
applied. Also, Proceedings of the Japan Society of Mechanical Engineers (C edition), Vol. 65, no.
637, Sano et al., "Study on modeling of acoustic transfer system for active noise control and
design method of PID feedback control system," pp. As shown in 121-127, a case where the
Smith method effective for a control object having a large time delay such as a chemical plant is
applied to the design of a feedback compensation circuit is also reported.
[0009]
However, in any case, the effect is limited to the resonance mode inside the narrow-band duct,
and there is a portion where the sound is increased in a part of the frequency range. That is, it is
necessary to trade off the gain in order to maintain the stability of the feedback system by the
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delay element as described above. If the gain is uniformly lowered in a wide frequency range, the
target gain can not be obtained at the target frequency, and a secondary sound that cancels out
the noise can not be generated. In addition, when the gain is adjusted for each frequency domain,
a phase difference occurs. Therefore, in order to obtain the noise reduction effect as the entire
noise frequency range, it is necessary to experimentally obtain the adjustment of the gain and the
phase difference for each frequency domain. Therefore, it has been very difficult to obtain
uniform noise reduction effects over a wide range of continuous frequencies.
[0010]
Also, the Journal of the Japan Society of Mechanical Engineers (C edition), Vol. 61, no. No. 588, et
al., "Active noise control of one-dimensional exhaust duct system by feedback control", pp. 588. It
is also pointed out at 118-124 that the noise reduction effect of the feedback type active noise
control device is mainly governed by the dynamic characteristics of the secondary sound
emitting speaker that emits the secondary sound.
[0011]
SUMMARY OF THE INVENTION The present invention has been made in view of the abovedescribed circumstances, and achieves both stability of the system and uniform noise reduction
in a wide range of continuous frequencies, and a system It is an object of the present invention to
provide a feedback type active noise control device capable of designing a typical controller.
[0012]
SUMMARY OF THE INVENTION In the feedback type active noise control system of the present
invention, the noise source is surrounded by a one-dimensional duct whose cross-sectional
dimension is sufficiently smaller than the wavelength of the noise to be silenced.
In such ducts, the sound waves of noise can be treated as plane waves.
And while providing the secondary sound radiation speaker which radiates | emits a secondary
sound in the duct, the secondary sound and noise radiated from a secondary sound radiation
speaker are more distant from the said noise source than a secondary sound radiation speaker
An error detection means is provided for detecting the sound pressure of the synthetic wave
generated by interference. Furthermore, the secondary sound generation filter performs
electrical signal processing on the sound pressure detected by the error detection means to
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generate a signal for secondary sound and outputs the signal to the secondary sound radiation
speaker. Thus, a feedback circuit is configured.
[0013]
In such a feedback circuit, in order to obtain a uniform noise reduction effect in a predetermined
frequency range, if the amplitude characteristic of the loop transfer characteristic G0 (s) of the
feedback circuit is constant in the predetermined frequency range Good. However, since the
frequency response characteristic P (s) from the input signal of the secondary sound radiation
speaker to the error detection means is included in G0 (s), its amplitude characteristic is not
constant. In general, the amplitude characteristic of G0 (s) can be made constant by multiplying
the inverse characteristic of P (s) (the inverse of the amplitude characteristic and the sign
conversion of the phase characteristic). However, P (s) includes a time delay until the sound wave
emitted from the speaker reaches the error detection means. That is, in order to obtain P (s),
when using the signal obtained by the error detection means, the sound signal including noise on
the cone surface of the secondary sound radiation speaker at the time when the sound reaches
the error detection means It will be necessary. The noise signal on the surface of the cone of this
secondary sound radiation speaker is a signal of the sound to be detected in the future by the
error detection means, so detection by the error detection means to obtain the inverse
characteristic of P (s) It will require the future value of the signal and can not be realized.
[0014]
Therefore, first, the dynamic characteristic of P (s) is separated into a minimum phase portion
Pmin (s) containing no delay component in a predetermined frequency range and the other
portion, and the inverse characteristic of this Pmin (s) is obtained. Design the secondary sound
generation filter to have. As a result, since the loop transfer characteristic G0 (s) of the feedback
circuit has a constant gain in a predetermined frequency range, a uniform noise reduction effect
can be obtained. Moreover, such a filter can be easily designed from the characteristics of the
secondary sound radiation speaker and the like, and can be realized without repeating
experimental trial and error.
[0015]
However, with this as it is, in the high frequency component, the phase difference becomes large
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due to the influence of the delay, and not only the noise can not be eliminated but also the
feedback system may become unstable. Therefore, the gain of the high frequency component is
reduced, and the influence of the high frequency component having a large phase difference is
suppressed. As a result, it is possible to avoid the risk of oscillation of the feedback system and to
perform stable noise control. The high frequency component of noise may be separately dealt
with by a sound absorbing material or the like.
[0016]
The secondary sound generation filter as described above includes, for example, constant
multiplication signal processing means for multiplying the electric signal of the sound pressure
detected by the error detection means by a constant, and inverting the output signal from the
constant multiplication signal processing means Inverting means, inverse characteristic signal
processing means for performing signal processing on the output signal from the inverting
means, low frequency pass signal processing means for reducing high frequency components of
the output signal of the inverse characteristic signal processing means, and It can consist of The
inverse characteristic signal processing means is, as described above, an amplitude characteristic
that has an inverse relationship with the amplitude characteristic of the electro-acoustic
frequency response from the input signal of the secondary sound radiation speaker to the output
signal of the error detection means within a predetermined frequency range. Can be configured
with an IIR digital filter. The numerator polynomial of the IIR digital filter used at this time
matches the denominator polynomial in the case where the electro-acoustic frequency response
characteristic is modeled as the IIR digital filter. Also, among the characteristic roots of the
numerator polynomial in the case where the electro-acoustic frequency response characteristic is
modeled as an IIR digital filter, the denominator polynomial is a characteristic root that is the
inverse of the characteristic root whose size exceeds 1.0. And match a polynomial that has the
same thing for characteristic roots whose size does not exceed 1.0.
[0017]
In the IIR digital filter of such a characteristic, since all characteristic roots in the denominator
polynomial do not exceed 1.0, it is stable if there is no delay component. Also, in this IIR digital
filter, among the characteristic roots of the molecular polynomial when the electro-acoustic
frequency response characteristic is modeled as an IIR digital filter, the inverse of the
characteristic roots with a magnitude of the characteristic root of more than 1.0 Give a
characteristic root. The gain is compensated when the inverse characteristic root is given in this
way. Further, the phase has a minimum phase characteristic excluding the delay component.
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[0018]
Such an IIR digital filter has an inverse characteristic of the minimum phase characteristic in a
predetermined frequency range of the electro-acoustic frequency response from the input signal
of the secondary sound radiation speaker to the output signal of the error detection means. ,
Second phase signal with minimum phase characteristics can be generated. Under this condition,
the phase difference increases as the frequency increases due to the delay component. Therefore,
the high frequency component of the output signal of the inverse characteristic signal processing
unit is reduced by the low frequency pass signal processing unit, and feedback is performed by
the high frequency component. It prevents the circuit from becoming unstable.
[0019]
DESCRIPTION OF THE PREFERRED EMBODIMENTS FIG. 1 is a block diagram showing an
embodiment of a feedback type active noise control system according to the present invention. In
the figure, 1 is a noise source, 2 is a duct, 3 is a secondary sound emission speaker, 4 is an error
detection unit, 5 is a secondary sound generation filter, 6 is a fixed signal processing unit, 7 is a
reverse unit, 8 is a reverse A characteristic signal processing unit 9 is a low frequency pass signal
processing unit.
[0020]
The duct 2 is formed so as to surround the noise source 1, and the longest length of its cross
section is made sufficiently shorter than the wavelength of the generated noise. In the duct of
such a configuration, the noise emitted from the noise source 1 can be approximated as a plane
wave.
[0021]
A secondary sound radiation speaker 3 for emitting secondary sound is disposed at a position on
the outlet side of the duct 2 as viewed from the noise source 1. The distance from the secondary
sound radiation speaker 3 to the outlet of the duct 2 may have a length (about several
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centimeters) such that a sound wave generated from the secondary sound radiation speaker 3
can be approximated as a plane wave. As described above, by making the longest length of the
cross section of the duct 2 sufficiently smaller than the noise wavelength, it is possible to treat all
the sound waves inside the duct 2 as plane waves, so the secondary sound radiation speaker 3
The silencing effect realized in the vicinity can be obtained uniformly in the cross section of the
duct 2.
[0022]
An error detection unit 4 is provided in the vicinity of the secondary sound radiation speaker 3.
The error detection unit 4 detects the sound pressure of the synthetic sound in which the noise
from the noise source 1 and the secondary sound from the secondary sound radiation speaker 3
interfere with each other. For example, it is comprised including a microphone etc.
[0023]
A secondary sound generation filter 5 is disposed downstream of the error detection unit 4. The
secondary sound generation filter 5 is composed of a constant-magnification signal processing
unit 6, an inverting unit 7, an inverse characteristic signal processing unit 8, and a low frequency
band pass signal processing unit 9. The fixed magnification signal processing unit 6 multiplies
the signal of the error detection means by a constant. The inverting unit 7 inverts the output
signal from the constant-magnification signal processing unit 6. The inverse characteristic signal
processing unit 8 has a relationship between the amplitude characteristic of the electro-acoustic
frequency response from the input signal of the secondary sound emission speaker 3 to the
output signal of the error detection unit 4 and the inverse of the minimum phase characteristic
within a predetermined frequency range. Signal processing is performed on the output signal
inverted by the inverting unit 7. The inverse characteristic signal processing unit 8 can be
configured by, for example, an IIR digital filter as described later. The low frequency range pass
signal processing unit 9 reduces high frequency components of the output signal of the inverse
characteristic signal processing unit 8.
[0024]
The signal detected by the error detection unit 4 is multiplied by a constant by the constant
multiplication signal processing unit 6 by the secondary sound generation filter 5 having such a
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configuration, and is further multiplied by -1 by the inversion unit 7 to be inverted. And sent to
the inverse characteristic signal processing unit 8. The inverse characteristic signal processing
unit 8 performs the signal processing according to the above-mentioned characteristic, and the
low frequency range pass signal processing unit 9 reduces high frequency components. Then, the
secondary sound radiation speaker 3 is driven by the output signal of the low frequency range
pass signal processing unit 9 to radiate the secondary sound into the duct 2.
[0025]
Here, a method of realizing the inverse characteristic signal processing unit 8 will be described in
detail below. First, the electro-acoustic frequency response from the input signal of the secondary
sound emission speaker 3 to the output signal of the error detection unit 4 is measured using an
FFT analyzer or the like. FIG. 2 is a graph showing an example of the electro-acoustic frequency
response from the input signal of the secondary sound radiation speaker 3 to the output signal of
the error detection unit 4. In the frequency response characteristic shown in FIG. 2, for example,
there is a peak of gain at a frequency of 100 tens of Hz, and the phase is inverted by 180
degrees. Such a large peak of gain often appears as a characteristic of the secondary sound
emission speaker 3. If a gain characteristic as shown in FIG. 2A is obtained, basically a signal
using this inverse characteristic is generated, and noise can be eliminated if interference occurs.
However, as described above, in order to ensure the stability of the feedback system, the gain can
not be 1 or more at a frequency at which the phase is reversed. That is, if the phase has a 180degree inverted characteristic, when the secondary sound of that frequency is emitted, it acts to
intensify the noise, which may cause oscillation. However, if the peak of such a large gain is
suppressed to 1 or less and the gain is uniformly reduced in all frequency ranges, there is a
possibility that the desired gain can not be obtained even at the target noise frequency.
[0026]
In the electro-acoustic frequency response from the input signal of the secondary sound emission
speaker 3 to the output signal of the error detection unit 4 as shown in FIG. 2, the sound radiated
from the secondary sound emission speaker 3 is A delay component until reaching the error
detection unit 4 is included. Therefore, a large change is particularly shown in the phase
characteristics shown in FIG. In the present invention, the inverse characteristic signal processing
unit 8 is configured to perform signal processing in the minimum phase characteristic except for
the delay component which has a great influence on such phase characteristic.
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[0027]
As shown in FIG. 2, when the electro-acoustic frequency response from the input signal of the
secondary sound radiation speaker 3 to the output signal of the error detection unit 4 is
obtained, the frequency response characteristic is numerically expressed in the form of an IIR
digital filter Model. The frequency response characteristic shown in FIG. 2 can be expressed as a
12th-order model when expressed in an IIR digital model.
[0028]
In general, the IIR digital model is represented by the ratio of the numerator polynomial to the
denominator polynomial (fractional equation) as follows. In this equation, the characteristic root
of the numerator polynomial is called "zero point", and the characteristic root of the denominator
polynomial is called "pole", and it is known as an important factor that determines the
characteristic of the digital filter. In addition, when the size of these "zeros" and "poles" exceeds
1.0, they are referred to as "unstable zeros" and "unstable poles", respectively. In particular, it is
known that the output of the digital filter diverges when there is even one "unstable pole".
[0029]
FIG. 3 is a distribution diagram of poles and zeros when the electro-acoustic frequency response
characteristic from the input signal of the secondary sound radiation speaker 3 to the output
signal of the error detection unit 4 is represented as an IIR digital model. The poles and zeros in
the IIR digital model obtained from the electro-acoustic frequency response from the input signal
of the secondary sound radiation speaker 3 shown in FIG. 2 to the output signal of the error
detection unit 4 are determined and plotted on the real-imaging plane It becomes like FIG. In FIG.
3, points indicated by black rhombus indicate poles, and points indicated by black triangles
indicate zeros. As can be seen with reference to FIG. 3, it can be seen that the poles are all within
a unit circle centered at the origin of radius 1.0. However, it can be seen that unstable zeros exist
at coordinates (1.01, 0) and coordinates (5.98, 0) in the case of zeros.
[0030]
In this IIR digital model, it is stable because there is no unstable pole. However, in order to
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generate a secondary sound that actually cancels out noise, an inverse characteristic of the
electro-acoustic frequency response from the input signal of the secondary sound radiation
speaker 3 to the output signal of the error detection unit 4 is required. That is, an inverse IIR
digital model in which the denominator polynomial and the numerator polynomial in the abovedescribed IIR digital model are interchanged is required. At this time, in the inverse IIR digital
model in which the denominator polynomial and the numerator polynomial are simply
interchanged, the unstable zero point in the original IIR digital model becomes an unstable pole,
which results in divergence.
[0031]
Therefore, in the IIR digital model described above, among the characteristic roots of the
numerator polynomial, for those whose size exceeds 1.0, the characteristic roots to be the inverse
number are given, and the size is 1.0 A new IIR digital model is designed to match a polynomial
that has the same one for characteristic roots that do not exceed. That is, the numerator
polynomial with the unstable zero point (1.01, 0) shown in FIG. 3 as (0.99, 0) and the unstable
zero point (5. 98, 0) as (0.167, 0) is shown. Ask. FIG. 4 is a distribution diagram of poles and
zeros in the new IIR digital model. The representation of the points in the figure is the same as in
FIG. By giving a characteristic root which is the reciprocal of the unstable zero as described
above, the zeros of the new IIR digital model are all within the unit circle as shown in FIG.
[0032]
As described above, the amplitude characteristic with respect to the frequency among the
frequency response characteristics of the original IIR digital model is stored as it is in the new IIR
digital model by the conversion operation giving the characteristic root which is the reciprocal of
the unstable zero point. Ru. The phase characteristic is realized as a so-called "minimum phase
characteristic" obtained by removing the time delay from the phase characteristic of the original
IIR digital model. Thus, in the original IIR digital model obtained from the electro-acoustic
frequency response from the input signal of the secondary sound emission speaker 3 to the
output signal of the error detection unit 4, the unstable zero point of the molecular polynomial is
its inverse The simple operation of giving characteristic roots gives the minimum phase
characteristic excluding the time delay component.
[0033]
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The inverse IIR digital model created by replacing the numerator polynomial and the
denominator polynomial of the new IIR digital model obtained as described above has no
unstable pole and is stable. Also, in this inverse IIR digital model, signals can be processed
without the need for future values. The inverse characteristic signal processing unit 8 can be
configured by an IIR digital filter that realizes such an inverse IIR digital model.
[0034]
As the frequency response of the portion combining the inverse characteristic signal processing
unit 8 and the secondary sound radiation speaker 3 realized as described above, the amplitude
characteristic is 1.0 in a predetermined frequency region and the phase becomes higher as the
frequency becomes higher. There remains a time delay component in which the phase angle is
delayed. This is because the inverse characteristic signal processing unit 8 separates the
minimum phase characteristic and the time delay component and performs signal processing on
the minimum phase characteristic, so the time delay component remains as it is.
[0035]
For example, if the distance from the secondary sound radiation speaker 3 to the error detection
unit 4 is about 3 cm, the phase delay at 1 kHz can be suppressed to 36 degrees, but at 5000 Hz
the phase is near 180 degrees. And become unstable as a system. However, in the high frequency
region of 5000 Hz, a passive noise reduction method using a sound insulating material or a
sound absorbing material is more effective than such active noise control. Therefore, in the
present invention, a low frequency band pass signal processing unit 9 for suppressing the gain in
the high frequency band is provided, and the noise is reduced by the passive noise reduction
method without performing the active noise control in the high frequency band. There is.
[0036]
FIG. 5 is a graph showing an example of the frequency response characteristic of the low
frequency pass signal processor. FIG. 5A shows gain characteristics, and FIG. 5B shows phase
characteristics. As shown in FIG. 5A, the low frequency range pass signal processing unit 9
reduces the gain in the high frequency range. This prevents oscillation or the like due to the
secondary sound having a large phase difference. In addition, since the phase adjustment is not
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performed in the low frequency range pass signal processing unit 9, as shown in FIG. 5B, the
phase is gradually delayed in the high frequency range. Of course, processing may be performed
to advance the phase as the frequency becomes higher, and it may be configured to correct the
phase delay even by a small amount.
[0037]
FIG. 6 is a Nyquist diagram showing the loop transfer characteristic of the system depending on
the presence or absence of the low frequency range signal processing unit. The example shown
in FIG. 6 shows an example in which the constant-magnification signal processing unit 6 is given
an amplitude of 1.2 times. In this case, when the low frequency pass signal processing unit 9 is
not provided, the loop transfer characteristic passes through the left side of the point of (-1.0, 0)
when passing the real axis. However, by providing the low frequency range pass signal
processing unit 9, the gain in the high frequency range is suppressed, and as shown in FIG. It can
be increased. This makes it possible to avoid the risk of oscillation.
[0038]
By configuring the secondary sound generation filter 5 as described above, stability of the system
and noise reduction in a continuous frequency range can be achieved. Further, each of the parts
constituting such a secondary sound generation filter 5 can be determined almost uniquely if the
characteristics of the secondary sound radiation speaker 3 are determined, and it is It can be
designed without relying on intuition or experience. In this example, the secondary sound
generation filter 5 is shown by four components, but it is also possible to configure some of them
as long as the above-mentioned function is achieved.
[0039]
FIG. 7 is a graph showing an example of the noise reduction effect by the conventional feedback
system, and FIG. 8 is a graph showing an example of the noise reduction effect according to the
present invention. In the figure, the thin solid line indicates the gain before silencing, and the
thick solid line indicates the gain after silencing. Furthermore, the broken line indicates the
feedback gain. In the conventional feedback system, as shown in FIG. 7, the feedback gain
(broken line) provided varies with frequency due to the influence of the characteristics of the
secondary sound radiation speaker, and the reduction effect is increased to a high frequency.
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Frequency is mixed. As a result, assuming that the predetermined frequency range is 1 kHz, for
example, when viewed as the noise reduction amount of the entire 1 kHz, only a noise reduction
effect of less than 2 dB can be obtained.
[0040]
On the other hand, in the noise reduction according to the present invention, as shown by a
broken line in FIG. 8, uniform feedback gain can be added over all frequencies. By this, it was
possible to realize noise reduction in a continuous frequency range. For example, when the
predetermined frequency range is 1 kHz, a noise reduction effect of 6 dB can be obtained as the
noise reduction amount of the entire 1 kHz.
[0041]
As is apparent from the above description, according to the feedback type active noise control
device of the present invention, stable noise reduction in a continuous frequency range can be
made possible. Furthermore, since there is no trial and error part in the controller design, it is
possible to greatly reduce development costs and mass production costs. As a result, there is an
effect that it is possible to provide a low-cost, very small active noise control device utilizing the
advantage of the feedback type.
[0042]
Brief description of the drawings
[0043]
FIG. 1 is a block diagram showing an embodiment of a feedback type active noise control system
according to the present invention.
[0044]
FIG. 2 is a graph showing an example of an electro-acoustic frequency response from an input
signal of a secondary sound emission speaker to an output signal of an error detection unit.
[0045]
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FIG. 3 is a distribution diagram of poles and zeros when an electro-acoustic frequency response
characteristic from an input signal of the secondary sound emission speaker to an output signal
of the error detection unit is represented as an IIR digital model.
[0046]
FIG. 4 is a distribution diagram of poles and zeros in the new IIR digital model.
[0047]
FIG. 5 is a graph showing an example of the frequency response characteristic of the low
frequency pass signal processing unit.
[0048]
FIG. 6 is a Nyquist diagram showing the round trip transfer characteristic of the system
depending on the presence or absence of the low frequency pass signal processing unit.
[0049]
FIG. 7 is a graph showing an example of the noise reduction effect by the conventional feedback
system.
[0050]
FIG. 8 is a graph showing an example of the noise reduction effect according to the present
invention.
[0051]
FIG. 9 is a schematic configuration view showing an example of a headphone using a
conventional feedback type active noise control device.
[0052]
10 is a block diagram showing an example of a general feedback system.
[0053]
Explanation of sign
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[0054]
DESCRIPTION OF SYMBOLS 1 ... noise source, 2 ... duct, 3 ... secondary sound radiation speaker, 4
... error detection part, 5 ... secondary sound generation filter, 6 ... fixed size signal processing
part, 7 ... inversion part, 8 ... reverse characteristic signal processing 9, low frequency pass signal
processing unit, 11 headphones, 12 diaphragms, 13 microphones.
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