JP3439228B2 - Noise canceling device - Google Patents

Description

DETAILED DESCRIPTION OF THE INVENTION [0001] BACKGROUND OF THE INVENTION 1. Field of the Invention The present invention relates to a noise canceling device.
In particular, the noise at the noise cancellation point (observation point)
High S / N ratio and high dynamic sound synthesized with cancel sound
Noise cap that can be detected in the range and input to the adaptive signal processor
The present invention relates to a cancel device. [0002] 2. Description of the Related Art As a countermeasure against noise, a sound absorbing material is conventionally used.
(Passive control) is known. But sucking
In the method using sound material, a quiet area with low noise is formed.
Is difficult to perform and bass cannot be effectively eliminated.
There is a title. In particular, to prevent noise in the cabin of a car
Increases the weight of the car and effectively reduces noise.
There is a problem that can not be done. As a result, the noise capa
Method to reduce noise by emitting cancel sound from speaker
(Active control) is in the limelight, such as factories and offices
Is being put to practical use in some indoor spaces. In addition,
Active control reduces noise even in the passenger compartment
A scheme has been proposed. FIG. 2 shows a conventional device for realizing noise cancellation.
FIG. 11 is a configuration diagram of an installation, where 11 is an engine that is a noise source,
Is a rotation speed sensor for detecting the engine rotation speed R, and 13 is an air speed sensor.
Constant amplitude sine having a frequency corresponding to the engine speed R
Wave signal to reference signal x_a1nReference signal generator generated as
It is. If the noise source is an engine,
The generated noise has periodicity (periodic noise),
The frequency depends on the engine speed. For example, 4-cylinder d
In the case of an engine, the periodic noise generated in the cabin is engineered.
The second harmonic of the rotation speed is dominant, and the rotation speed is 600
At 10 rpm, the noise generated in the cabin
The frequency is 20 Hz, and the rotation speed is 6000 rpm (100 rpm).
ps), the frequency of the noise generated in the vehicle interior is 200
Hz. The reference signal generating unit 13 outputs a positive second harmonic signal.
Sine wave data is stored in ROM, and that data is needed
Is read and output according to the reference signal x._a1n
Generate In addition, read / output timing of this data
Is controlled according to the engine speed R.
Noise generated according to the engine speed R
A reference signal having a frequency of
You. [0004] 14 is a noise canceling controller.
The reference signal x generated from the reference signal generator 13_a1nTo
As well as the noise cancellation position in the vehicle interior (observation
Point Sn, for example, near the driver's ear)₁When
Cancel sound Sc₁Error signal e_1nAs
Adaptive signal processing so that the error signal is minimized.
Noise canceling signal y_a1nIs output. noise
The cancel controller 14 includes an adaptive signal processing unit 14a
An adaptive filter 14b having a digital filter configuration;
Cancellation sound propagation system from peaker to noise cancellation point
Created based on the transfer function of (secondary sound propagation system) and referenced
Signal x_a1nIs input to the filter 14c.
15 is an adaptive filter output (noise canceling signal y_a1n)
Converter that converts the signal into an analog noise cancellation signal
And 16 are power amplifiers for amplifying the noise canceling signal.
17 is the noise canceling sound Sc₁Radiating cancel
The speaker 18 is located at the noise canceling point, and the noise S
n₁And cancel sound Sc₁Detects the synthesized sound of
Error signal e_1nError microphone, and 19 is
Signal e_1nAmplifier 20 amplifies the periodic noise
Low-pass filter for removing out-of-band noise signals, 20 '
Is an AD converter that converts the low-pass filter output to digital
It is Bata. [0005] The adaptive signal processing section 14a has a noise canceling point.
Error signal e at_1nAnd input via filter 14c
Reference signal R for signal processing₁₁Is entered and these
The noise at the noise cancellation point using the signal
Adaptive signal processing so that the adaptive filter 14b
Determine the coefficient. For example, the adaptive signal processing unit 14a is a well-known
According to LMS (Least Mean Square) adaptation algorithm
The error signal e input from the error microphone 18_1nBut
Determine the coefficient of the adaptive filter 14b so as to minimize it
You. The adaptive filter 14b is controlled by the adaptive signal processing unit 14a.
The reference signal x according to the determined coefficient_a1nDigital filter
Noise reduction from DA converter 15
Signal y_a1nIs output. Note that the reference signal x_a1nErase
Want noise Sn₁Signals must be highly correlated with
Sounds that have no correlation with the illumination signal are not eliminated. Engine 11
When rotated, the rotation speed R is detected by the rotation speed sensor 12.
And the reference signal generating unit 13 responds according to the engine speed R.
Reference signal x_a1n(See Fig. 3 (a))
Input to the sound cancel controller 14. At this time,
Engine sound (peripheral) generated from the engine 11
Periodic noise) is a noise propagation system having a predetermined transfer function.
Noise cancellation by propagating through the air with (primary sound propagation system)
To the point. Therefore, the noise at the noise cancellation point
(Engine sound) Sn₁Is slightly weaker and slightly slower
As shown in FIG. First, the noise cancellation controller 14
For example, the reference signal x_a1nNoise cancellation signal with opposite phase
y_a1nIs output from the cancel speaker 16 as shown in FIG.
Show cancel sound Sc₁Is output. However, the noise Sn₁of
Since the level and the phase are shifted, the cancel sound Sc₁By
Noise is not canceled and the error signal e_1nOccurs
You. The noise cancellation controller 14 receives the error signal e.
_1nAdaptive signal processing is performed to minimize
The coefficient of the data 14b is determined.
As shown in (d), the cancel sound Sc₁The phase of the noise Sn₁of
Cancel the noise by making the phase opposite to the phase and matching the level
To For the sake of simplicity, the noise source is described above.
One cancel sound source (speaker)
This is an example in which one cancel point (observation point) is set. Only
However, there are actually multiple noise sources, and
There are several points (observation points) that we want to
The speaker cannot cancel the noise at each observation point,
There are multiple manufacturers. Figure 4 shows K noise sources and speakers
Conventional noise canceling with M observation points and L observation points
It is a block diagram of a cell apparatus. 21 is the noise at each observation point
DSP (Digital System) that operates to cancel
Noise canceling control with a signal processor
La and 22 indicate the noise from each noise source (not shown) to each observation point.
A primary sound virtual propagation system (a noise propagation
), 23 are each from each speaker, including the characteristics of the speakers.
Second order representing the system in which the cancellation sound propagates to the observation point
Sound propagation system (cancellation sound propagation system), 24 at each observation point
A signal synthesizer that expresses the function of a microphone that
₁~ 24₁'Corresponds to the microphone at the first observation point,
Part 24_Two~ 24_Two'Corresponds to the microphone at the second observation point
..., Adding section 24_L~ 24_L'At the L-th observation point
Equivalent to a microphone. d_d1n~ D_dLnIs the key at each observation point.
External noise not subject to cancellation. In addition, DA converter
And an AD converter are omitted. The noise canceling controller 21 is a DSP
And a multi-input / multi-output adaptive filter (hereinafter simply referred to as an adaptive filter).
21a, and filtered X signal creation
Filter 21b and the adaptive signal processing unit 21c.
It is. The adaptive filter 21a includes a reference signal generator (not shown).
Reference signal x input from_a1n~ X_aKnTo the specified file
Noise canceling signal y after filtering_a1n~ Y
_aMnAnd the noise cancel signal is input to each speaker.
Power. Filter 21b for creating filtered X signal
Illumination signal x_a1n~ X_aKnThe transfer function mat of the secondary sound propagation system 23
For signal processing by convolving each element of Rix (propagating element)
Reference signal (filtered X signal) r_111n~ R_LMKnOut
Power. The adaptive signal processing unit 21c detects the error at each observation point.
-Signal e_1n~ E_LnAnd the filter output from the filter 21b.
Lutard X signal r_111n~ R_LMKnAre input, these signals
To cancel noise at each observation point
Performs adaptive signal processing to determine coefficients of adaptive filter 21a
I do. FIG. 5 is an explanatory diagram of the primary sound virtual propagation system 22.
As shown in FIG. 5 (a), the K noise sources NG₁~ NG_K
The noise generated from the noise has a certain frequency / phase characteristic.
Microphones (M
IC₁~ MIC_L). Therefore, the ith noise
The noise from the source NGi is the jth microphone MIC_jThe story that reaches
Set the transfer characteristics of the transport system to H_jiThen, the primary sound virtual propagation system 22
Is expressed as shown in FIG. 5 (b), and its transfer function matrix
(H) is as follows. [0010] (Equation 1) Each element H of the transfer function matrix (H)_ijIs
Modeled by the FIR digital filter shown in FIG.
It is. That is, the input signal is sequentially delayed by one sampling time.
Delay element DL to be extended and a coefficient h₀,
h₁, H_Two, Multiplying unit ML, and multiplying unit output
Digital filter consisting of an adder AD that adds
Is converted to FIG. 7 is an explanatory diagram of the secondary sound propagation system 23,
As shown in FIG. 7A, each speaker SP₁~ SP_MOriginated from
Noise canceling sound has predetermined frequency and phase characteristics
Microphones provided at each observation point by propagating through the secondary sound propagation system 23
MIC₁~ MIC_LTo reach. Therefore, the ith noise
Cancel signal y_ainJ-th cancellation sound based on
Microphone MIC_jThe transfer characteristic of the secondary sound propagation system up to_ji
In this case, the secondary sound propagation system 23 is configured as shown in FIG.
And its transfer function matrix (C) is
Swell. [0012] (Equation 2) Each element of the transfer function matrix (C) is linear
As in the case of the sound virtual propagation system 22, the FIR type shown in FIG.
Modeled by a digital filter. That is,
A delay element DL for sequentially delaying a force signal by one sampling time
And a coefficient c for each delay element output ₀, C₁, C_Two, ...
Multiplying unit ML for performing the calculation and an adding unit A for adding the outputs of the respective multiplying units
It is modeled by a digital filter consisting of D. FIG. 8
Each element of the transfer function matrix (C) of the secondary sound propagation system 23
C_ijFor creating a filtered X signal created using
It is a block diagram of the data 21b. Refer to the adaptive signal processing unit 21c
Signal x_a1n~ X_aKnAnd noise and cancellation at each observation point
Signal (error signal) e_1n~ E_LnAnd based on
Perform adaptive signal processing to update adaptive filter coefficients
The adaptive filter 21a outputs the reference signal x_a1n~ X_aKnEnter
Noise cancel signal y_a1n~ Y_aMnSpies
Input to the vehicle and cancel the noise at each observation point. The output from the adaptive filter 21a is
Noise cancel signal y_a1n~ Y_{a Mn}Is the observation point
Frequency and phase of the secondary sound propagation system 23
Reached under the influence of characteristics. For this reason, adaptive signal processing
The unit 21c outputs the reference signal x_a1n~ X_aKnJust use
And a signal obtained by adding the characteristics of the secondary sound propagation system 23 to the reference signal.
X LMS (MEFX LMS)
Adopting algorithm, more advanced noise cancellation control
Is going. That is, the filtered X LMS algorithm
In the rhythm, the reference signal x_a1n~ X_aKnTo the filter 21b
More filtered signal (filtered X signal)
Adaptive filter 2 using error signal at each observation point
1a is updated. In FIG. 8, C_ijIs the secondary sound propagation system 23
Element C of transfer function matrix (C) in_ij(FIG. 7
FIR type digital filter that realizes the above-mentioned function. H
The filter 21b receives each reference signal x_a1n~ X_aKnTo all propagation
Convolve the elementary characteristics (the filter corresponding to all
Filter X signal r)_111n~ R_{LM Kn}
Is output. That is, the reference signal x
_a1nThe propagation element from the first speaker to all observation points
C₁₁~ C_L1To make the filtered X signal r_{11 1n}~ R
_L11nAnd outputs a reference signal x_a1nIs the second speaker
Element C to all observation points₁₂~ C_L2Act
Lutard X signal r_121n~ R_L21nIs output, ...
No. x_a1nFrom the Mth loudspeaker to all observation points
Element C_1M~ C_LMTo make the filtered X signal r_1M1n
~ R_LM1n, And similarly, the reference signal x_a2n~ X_aKn
To all the propagation elements. still, R₁₁= (R_111n, R_211n, ... r_L11n) R_{twenty one}= (R_121n, R_221n, ... r_L21n) ... R_M1= (R_1M1n, R_2M1n, ... r_LM1n) ... R_MK= (R_1MKn, R_2MKn, ... r_LMKn) Is expressed as FIG. 9 shows a multi-input / multi-output adaptive filter 21.
3A is a configuration diagram of a primary sound virtual propagation system 22 and a secondary sound propagation
It has the same structure as the system 23. A_11n~ A_MKnIs FI
It is composed of an R-type digital filter.
Delay elements DL1, Dl that are sequentially delayed by one sampling time
2 and the coefficient a for each delay element output₀, A₁, A_Two・ ・
Multiplying units ML1, ML2, ML3,.
Are realized by the adders AD1, AD2,.
You. The number of delay stages is not limited to two. Each reference signal x_a1n
~ X_aKnTo digital filter A_11n~ A_1KnAnd enter
Noise input to the first speaker
Cancel signal y_a1nAnd each reference signal x_a1n~ X_aKn
To digital filter A_21n~ A_2KnAnd add
Cancels the noise input to the second speaker
Signal y_a2n... Each reference signal x_a1n~ X_aKn
To digital filter A_M1n~ A_MKnAnd add
Cancels the noise input to the Mth speaker
Signal y_aMnIs obtained. Each FIR type data in the adaptive filter 21a
Digital filter A_11n~ A_MKnBy three coefficients (two-stage delay)
, The adaptive signal processing unit 21 c
Tal filter A_11n~ A_MKnAdaptive signal processing for each of the three coefficients
To determine the coefficient value. That is, one FIR
Type digital filter A_ijCoefficient a of₀, A₁, A_Twoabout
The coefficient a is calculated using the coefficient update equation₀, A₁, A
_TwoTo determine. [0018] (Equation 3) In equation (1), (n) is the current sampling time
(N-1) is the value one sampling time before, and (n-2) is two samples
The value before the sampling time, (n + 1) is the next sample from the current time.
It means the value up to the switching time. Therefore, R_ij(n-2) is
Filter 21 corresponding to reference signal two sampling times before
b means output, R_ij(n-1) is one sampling time before
Filter output according to the reference signal, R_ij(n) is the current time
Is a filter output corresponding to the reference signal. Also, μ is adaptive
A value less than or equal to 1 that determines the step for updating the filter coefficients
Number (step size parameter)
It is set to an appropriate value according to the system. In addition,
The larger the value of the size parameter μ, the greater the
The number approaches the optimal value, the speed increases, and the followability improves.
However, overshoot occurs after approaching and the stability is low.
Down. Also, the value of the step size parameter μ is small
The closer to the optimum coefficient value, the lower the speed and the lower the tracking ability
However, the overshoot after approaching the optimal value is small and cheap.
The qualification becomes good. e_nIs the noise at each of the L observation points
And a cancel sound, and R_ij, E_nIs it
These are expressed as follows.   R_ij= (R_1ijn, R_2ijn, ... r_Lijn)   R_ij(n) = (C_1i, C_2i, C_3i, ..., C_Li) X_ajn(n)   R_ij(n-1) = (C_1i, C_2i, C_3i, ..., C_Li) X_ajn(n-1)   R_ij(n-2) = (C_1i, C_2i, C_3i, ..., C_Li) X_ajn(n-2) [0020] (Equation 4) According to such a noise canceling device,
The signal processing unit 21c is a filter which is an output of the filter 21b.
Tard X signal r_111n~ R_LMKnAnd the noise at each observation point
Sound signal (error signal) e of the sound and the cancel sound_1n~ E
_LnAdaptive signal processing is performed based on
Fa Digital Filter A Constituting 1a_11n~ A
_MKnAnd the adaptive filter 21a determines the coefficient of the reference signal x
_a1n~ X_aKnAnd the noise cancel signal y_a1n~ Y
_aMnGenerating speaker SP₁~ SP_MFill in (Fig. 7)
Each speaker generates a cancellation sound and the noise at each observation point
Acts to cancel the sound. FIG. 10 shows the number of noise sources K = 1 and the number of speakers M =
2. Specific noise when the number of observation points (number of microphones) L = 2
It is a configuration diagram of a cancellation device, for example, in front of a car
The engine sounds in the two seats (driver's seat and passenger seat)
It is used to cancel. 21a is 2
FIR digital filters A_11n, A_21nConsists of
Adaptive filter 21b is a transfer function mat of the secondary sound propagation system.
Rix (C) each propagation element C₁₁, C_{twenty one}, C₁₂, C_{twenty two}To
For creating a filtered X signal composed of digital filters
Filters, 21c-1 to 21c-2 are adaptive signal processing units (MEFX
LMS algorithm), SP ₁, SP_TwoIs below each seat
Speaker provided, MC₁, MC_TwoIndicates each observation point (occupant's ear
Microphone). Each adaptive signal processing
Section, adaptive filter, filter for creating filtered X signal
Is a single DSP (Digital Signal Processor)
(A). FIG. 11 shows the number of noise sources K = 1 and the number of speakers M =
4. Specific conventional cases when the number of observation points (number of microphones) L = 4
FIG. 2 is a configuration diagram of the noise canceling device of FIG.
This is to cancel the engine sound in the right 4 seats.
You. 21a has four FIR digital filters A_11n,
A_21n, A_12n, A_22nAdaptive filter composed of
b denotes each transfer of the transfer function matrix (C) of the secondary sound propagation system.
Carrying element C₁₁~ C₄₁, C₁₂~ C₄₂, C₁₃~ C₄₃, C₁₄~ C
₄₄X-signal generation with digital filter
And an adaptive signal processor (MEF).
X LMS algorithm), SP₁, SP_Two, SP_Three, S
P_FourIs speaker, MC₁, MC_Two, MC_Three, MC_FourIs the observation point
Microphone. Each adaptive signal processor, adaptive processor
Filter, the operation of the filter for creating the filtered X signal is 1
DSP (Digital Signal Processor)
Be executed. [0024] The above-described noise cancellation
In the device, the noise at the noise cancellation point and the cancellation
Amplifies the error signal, which is a synthesized sound with the sound, with a microphone amplifier
After that, convert to A / D and input to the noise canceling controller.
You. The gain of this microphone amplifier is in terms of S / N ratio.
It is desirable to set as high as possible, but if you increase the gain,
Exceeds the dynamic range of the AD converter
This causes distortion such as lip. For this reason, the gain is
Cannot be increased and a high-precision synthesized sound signal with a good S / N ratio
There is a problem that input cannot be made to the noise cancellation controller.
Was. From the above, an object of the present invention is that the S / N ratio is good,
Can also generate synthesized sound signals with a wide dynamic range
Noise canceling device that can be input to the
To provide. [0025] The above object is achieved by the present invention.
The noise cancellation controller to a digital signal
Processor (DSP) and gain available
We convert strange amplifier and amplifier output to digital data
AD converter for input to digital signal processor
Data and the input signal of the AD converter
The gain of the variable gain amplifier is controlled so as not to exceed the range.
Control and gain change of DSP scaling constant
Achieved by a controller that controls in the opposite direction
You. [0026] [Function] The controller receives the input signal of the AD converter.
Variable gain amplifier so that the dynamic range of
Control the gain of the DSP and scale the DSP.
The number is controlled in the direction opposite to the gain change direction. That is,
Controller instructs to multiply amplifier gain by 1 / n
Then, the numerical value input from the AD converter is
Is multiplied by n. This allows the microphone to
The gain G up to the response signal processing unit can be kept constant.
Can exceed the dynamic range even if the G is increased
No, high S / N ratio, high dynamic range synthesized sound signal
Can be input to the signal processing section of the noise cancellation controller.
You. [0027] FIG. 1 shows an embodiment of a noise canceling device according to the present invention.
It is a block diagram.Overall configuration In the figure, 31 is an engine which is a noise source, and 32 is an engine
A rotational speed sensor for detecting the rotational speed R, 33 is the engine rotational speed
The second harmonic sine wave signal corresponding to R is referred to as a reference signal x_a1nWhen
And a reference signal generator for outputting the reference signal. . 34 is inside the car
Check the engine sound at the noise canceling position (observation point).
Noise of DSP configuration that performs adaptive signal processing for canceling
A sound canceling controller, and a reference signal generating unit 33
Reference signal x generated from_a1nAnd the vehicle compartment
Noise Sn at the noise cancellation point in the building₁And cancel sound
Sc₁Error signal e_1nIs entered as
Perform adaptive signal processing to minimize error signals and
Sound cancel signal y_a1nIs output. Noise canceler
The controller 34 includes an adaptive signal processing unit 34a and a digital
The filter 34b and the speaker
Cancel sound propagation system up to the sound cancellation point (secondary sound propagation
System), the reference signal x_a1nBut
Filter 34c for creating a filtered X signal to be input
And the adaptive signal processing section multiplies the numerical data by n according to the instruction.
It has a scaling unit 34d to pass to 34a. 35 is an adaptive filter output (noise cancellation)
Signal y_a1n) To an analog noise cancellation signal
DA converter 36 amplifies the noise cancellation signal
Power amplifier 17 is a noise canceling sound Sc₁Emits
Cancellation speaker, 38 is located at the noise cancellation point
And the noise Sn₁And cancel sound Sc₁Detect the synthesized sound of
The synthesized sound signal is converted into an error signal e._1nEramai output as
, 39 is the error signal e_1nVariable gain my to amplify
Gain can be varied programmably
40 removes out-of-band noise signals of periodic noise
Low-pass filter, 41 digitally outputs low-pass filter output
Noise canceling con-
AD converter input to the controller, 42
G of the step 39 and scaling of the scaling unit 34d
It is a gain control unit for controlling the switching constant S. [0029]Adaptive signal processing unit The adaptive signal processing unit 34a detects an error at the noise cancellation point.
-Signal e_1nAnd the signal processing input through the filter 34c
Input reference signals for noise reduction and use these signals to reduce noise
Adaptive signal to cancel noise at the cancel point
Signal processing to determine the coefficients of the adaptive filter 34b.
For example, the adaptive signal processing unit 34a is a well-known LMS (Least Mea
n Square) according to the adaptive algorithm
Error signal e input from_1nTo minimize
The coefficient of the adaptive filter 34b is determined. Adaptive filter 34
b is in accordance with the coefficient determined by the adaptive signal processing unit 34a.
Reference signal x_a1nDigitally filtered to DA
Noise cancel signal y from converter 35_a1nOutput
You. [0030]Gain control section The gain control unit 42 outputs the output of the AD converter 41.
Force data is input, and an AD converter is
The input signal to the
Controlling the gain of the variable gain amplifier 39
And the scaling constant S of the DSP in the direction opposite to the gain change direction.
Control in the direction. For example, if the gain control
Do not exceed the converter output value and dynamic range
Various correspondences with the gain control variable n are set in advance.
In advance, gain control is performed based on the output value of the AD converter.
Find the variable n and multiply the gain of the amplifier 39 by 1 / n.
And AD conversion to the scaling unit 34d.
Is instructed to multiply the numerical value input from the data 41 by n.
As described above, the error signal from the error microphone 38 to the adaptive signal processing unit 34
a, the total gain G up to a can be kept constant.
Does not exceed dynamic range even if gain G is increased
I can do it. [0031]Overall behavior When the engine 31 rotates, its rotation speed R becomes a rotation speed sensor.
The reference signal generation unit 33 detects the
Reference signal x having a frequency corresponding to rotation speed R_a1nRaises the noise
The sound is input to the sound cancel controller 34. At this time,
Engine sound (peripheral) generated from the engine 31
Periodic noise) is a noise propagation system having a predetermined transfer function.
Noise cancellation by propagating through the air with (primary sound propagation system)
To the point. Noise and cancellation at the noise cancellation point
The synthesized sound (error signal) of the sound is detected by the error microphone 38.
And output to the variable gain amplifier 39. Gain available
The variable amplifier 39 outputs the error signal e._1nAmplify the low pass filter
The data is input to the AD converter 41 via the filter 40. still,
First, the gain of the variable gain amplifier 39 is set to the initial value G.
Have been. The AD converter 41 receives the input error signal.
Is converted to a digital signal and input to the scaling unit 34d.
You. Initially, the scaling constant S is 1, so
The digital data is directly input to the adaptive signal processing unit 34a.
You. In parallel with the above, for creating a filtered X signal
The filter 34c outputs the reference signal x_a1nIs entered and the LMS
Generate and adapt filtered X signals for adaptive signal processing
The signal is input to the signal processing unit 34a. The adaptive signal processing unit 34a
Using the error signal and the filtered X signal,
To perform the LMS adaptive signal processing, and
Determine the number. The adaptive filter 34b includes the adaptive signal processing unit 3
4a according to the coefficient determined by the reference signal x_a1nTo
Noise canceling signal y after digital filtering_a1n
Is input to the DA converter 35. DA converter 35
Is the noise cancel signal y_a1nTo DA conversion
Input to the speaker 37 via the loop 36. This allows
The noise canceling sound is output from the speaker 37 and the secondary sound is output.
The noise cancellation point is reached via the propagation system, and the noise is canceled.
Acts like a cell. Thereafter, the above operation is repeated.
The noise is canceled immediately. In the above, the error signal e_1nIs bigger
When the output value of the AD converter 41 increases,
The control unit 42 receives the input signal of the AD converter.
Variable gain amplifier so as not to exceed the dynamic range
Instruct 39 to reduce the gain to 1 / n,
The scaling unit 34d multiplies the scaling constant S by n.
To instruct. Thus, the variable gain amplifier 39
Error signal e_1nIs amplified by a gain G / n (<G).
Therefore, the amplified error signal is output from the AD converter 41
A / D conversion correctly without distorting without exceeding the mic range
Then, it is input to the noise cancellation controller 34.
Scaling unit 34 of noise cancellation controller 34
d is numerical data input with scaling constant S (= n)
Data to the adaptive signal processing unit 34a
input. Also, the error signal e_1nHas become smaller
When the output value of the inverter 41 decreases, the gain variable
Instructs the amplifier 39 to multiply the gain by m,
Reduce the scaling constant S to 1 / m
To instruct. From the above, adapt from the error microphone 38
The total gain G up to the signal processing unit 34a is always constant.
Even if this gain G is increased, dynamic
You can stay within the cleanse. In the above, based on the output of the AD converter
I controlled the gain and the scaling constant.
Control based on the output of the section and the output of the variable gain amplifier
You can also. As described above, the present invention has been described with reference to the embodiments.
However, the present invention follows the gist of the present invention described in the claims.
Various modifications are possible, and the present invention excludes these.
Not. [0035] As described above, according to the present invention, an AD converter
Make sure that the input voltage does not exceed its dynamic range.
In addition to controlling the gain of the variable amplifier,
It controls the calling constant in the direction opposite to the gain change direction.
Configuration from the microphone to the adaptive signal processor.
Gain G can be kept constant.
High dynamic range
S / N ratio, high dynamic range synthesized sound signal
It can be input to the signal processing section of the cancel controller.
Wear.

BRIEF DESCRIPTION OF THE DRAWINGS FIG. 1 is a configuration diagram of an embodiment of a noise canceling device of the present invention. FIG. 2 is a configuration diagram of a conventional noise canceling device. FIG. 3 is a waveform diagram for explaining a noise canceling operation. FIG. 4 is a configuration diagram of a conventional noise canceling device when a plurality of noise sources, speakers, and observation points exist. FIG. 5 is an explanatory diagram of a primary sound virtual propagation system. FIG. 6 is a configuration diagram of a digital filter that realizes each element of a transfer function matrix. FIG. 7 is an explanatory diagram of a secondary sound propagation system. FIG. 8 is a configuration diagram of a filter for creating a filtered X signal. FIG. 9 is a configuration diagram of an adaptive filter. FIG. 10 is a configuration diagram of a noise canceling device when one noise source, two speakers, and two microphones exist. FIG. 11 is a configuration diagram of a noise canceling apparatus in a case where there is one noise source, four speakers, and four microphones. [Description of Signs] 33 ··· Reference signal generating unit 34 ··· Noise cancel controller 34a ··· Adaptive signal processing unit 34d ··· Scaling unit 38 ··· Error microphone 39 ··· Gain variable amplifier 41 ··· AD converter 42 ··· Gain Control section

──────────────────────────────────────────────────続 き Continuation of the front page (72) Kunio Miyauchi 1-4-1 Chuo, Wako-shi, Saitama Prefecture Honda Technical Research Institute Co., Ltd. (56) References JP-A-4-77108 (JP, A) (58) Field surveyed (Int. Cl. ⁷ , DB name) G10K 11/178 F01N 1/00

Claims (1)

(57) [Claims] [Claim 1] A cancel sound source that outputs a cancel sound to cancel noise at a noise cancel point and a combined sound of the noise and the cancel sound at the noise cancel point are detected. A sensor, a synthesized sound signal at the noise cancellation point, and a reference signal corresponding to the noise generated from the noise source are input, and adaptive signal processing is performed using these signals to cancel the noise at the noise cancellation point, thereby canceling the noise. A noise canceling device including a noise canceling controller for generating a signal and inputting the signal to a canceling sound generating source, wherein the noise canceling controller is constituted by a digital signal processor and the synthesized sound signal is amplified.
A gain variable amplifier for the AD converter to be input to the digital <br/> Le signal processor converts the amplifier output into digital data, the input signal of the AD converter does not exceed the dynamic range of the AD converter So that the magnitude of the synthesized sound signal
And a controller that controls the gain of the variable gain amplifier based on the gain and controls the scaling constant of the digital signal processor in the direction opposite to the direction in which the gain changes.