JPH06130968A - Adaptive active muffling device - Google Patents

Adaptive active muffling device

Info

Publication number
JPH06130968A
JPH06130968A JP4281588A JP28158892A JPH06130968A JP H06130968 A JPH06130968 A JP H06130968A JP 4281588 A JP4281588 A JP 4281588A JP 28158892 A JP28158892 A JP 28158892A JP H06130968 A JPH06130968 A JP H06130968A
Authority
JP
Japan
Prior art keywords
noise
signal
coefficient
adaptive filter
detector
Prior art date
Legal status (The legal status is an assumption and is not a legal conclusion. Google has not performed a legal analysis and makes no representation as to the accuracy of the status listed.)
Pending
Application number
JP4281588A
Other languages
Japanese (ja)
Inventor
Satoshi Kawano
聖史 川野
Current Assignee (The listed assignees may be inaccurate. Google has not performed a legal analysis and makes no representation or warranty as to the accuracy of the list.)
Sanyo Electric Co Ltd
Original Assignee
Sanyo Electric Co Ltd
Priority date (The priority date is an assumption and is not a legal conclusion. Google has not performed a legal analysis and makes no representation as to the accuracy of the date listed.)
Filing date
Publication date
Application filed by Sanyo Electric Co Ltd filed Critical Sanyo Electric Co Ltd
Priority to JP4281588A priority Critical patent/JPH06130968A/en
Publication of JPH06130968A publication Critical patent/JPH06130968A/en
Pending legal-status Critical Current

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  • Soundproofing, Sound Blocking, And Sound Damping (AREA)
  • Circuit For Audible Band Transducer (AREA)

Abstract

PURPOSE:To provide the muffling device which is ready for the generation of noise at all times while providing a sufficient muffling effect in any frequency band of noise even when the coefficient of an adaptive filter is observed based on a time base. CONSTITUTION:This device is composed of a noise detector 1 for detecting noise generated from a noise source and outputting a noise signal, speaker 2 for muffling to generate an additional sound for arresting noise generated from the noise source, error detector 3 for detecting the synthetic sound of these noise generated from the noise source and additional sound and outputting an error signal, adaptive filter ADFi for preparing the driving signal of the speaker 2 by inputting the noise signal, coefficient calculation part LMSi for calculating the coefficient of the adaptive filter based on the noise signal and the error signal, and band pass filters FILi having the same characteristic and provided at positions between each of noise detectors and each of coefficient calculation parts respectively.

Description

【発明の詳細な説明】Detailed Description of the Invention

【0001】[0001]

【産業上の利用分野】本発明は適応フィルタと該適応フ
ィルタの係数を算出する係数演算部を有する適応能動消
音装置に関するものである。
BACKGROUND OF THE INVENTION 1. Field of the Invention The present invention relates to an adaptive active silencer having an adaptive filter and a coefficient calculator for calculating the coefficient of the adaptive filter.

【0002】[0002]

【従来の技術】従来より、騒音は大きな社会問題であ
り、生活空間における騒音を減少させたいという要望が
強かった。これまでは騒音レベルを減少させる方法とし
て内装材や遮音材を室内に装備したり、また騒音源の周
囲を金属やコンクリートなどの遮音材で包囲することな
どが行われてきたが、近年デジタル信号処理の発達によ
り逆位相の音による消音技術(適応消音システム)が利
用されるようになってきた。
2. Description of the Related Art Conventionally, noise has been a major social problem, and there has been a strong demand for reducing noise in living spaces. Until now, as a method to reduce the noise level, the interior materials and sound insulation materials have been installed in the room, and the noise source has been surrounded by sound insulation material such as metal or concrete. Due to the development of processing, the silencing technology (adaptive silencing system) by the sound of the opposite phase has come to be used.

【0003】ところで係る適応消音システムの制御部分
である適応制御部には主として図に示されるFiltered-X
というアルゴリズムが用いられていた。図 において1
01は騒音検出器としてのマイクロフォン、102は消
音用スピーカ、103はエラー検出器としてのマイクロ
フォン、104は前記騒音用マイクロフォン101によ
って検出された騒音信号に基づいて前記消音用スピーカ
102の駆動信号を出力する適応フィルタ(ADF)、
105は前記騒音信号と前記エラー検出用マイクロフォ
ンによって検出された騒音信号とスピーカ102からの
付加音とを合成して得られるエラー信号とを用いて前記
適応フィルタ104の係数を算出する係数演算部(LM
S)、106はスピーカ102から出力したエラー信号
がエラー検出用マイクロフォン103に到達するまでの
伝達特性である。
By the way, the adaptive control unit, which is the control unit of the adaptive muffling system, mainly includes the Filtered-X shown in the figure.
That algorithm was used. 1 in the figure
Reference numeral 01 is a microphone as a noise detector, 102 is a muffling speaker, 103 is a microphone as an error detector, and 104 is a driving signal for the muffling speaker 102 based on the noise signal detected by the noise microphone 101. Adaptive filter (ADF),
A coefficient calculation unit 105 calculates a coefficient of the adaptive filter 104 using an error signal obtained by combining the noise signal, the noise signal detected by the error detection microphone, and the additional sound from the speaker 102. LM
S) and 106 are transfer characteristics until the error signal output from the speaker 102 reaches the error detection microphone 103.

【0004】ところで上記の方法は騒音の全周波数帯域
を消音制御帯域としているにもかかわらず、実際は全帯
域の消音をするための時間が非常に多くかかり、且つ全
帯域を効率よく消音できないという問題点があった。
By the way, although the above-mentioned method uses the entire frequency band of noise as the muffling control band, in reality, it takes a very long time to muffle the entire band, and it is impossible to muffle the entire band efficiently. There was a point.

【0005】この種Filtered-Xアルゴリズムの問題点を
改善するための一つの方法として特開平3−28409
8号公報の技術が提案された。この技術のFiltered-Xア
ルゴリズムを基本とした概略構成を図 に示す。同図に
おいて前記図 と同じ構成要素については同一符号を付
してあるが、この図の場合、騒音検出用のマイクロフォ
ン101から適応フィルタ1041あるいは1042に
騒音信号が伝播される前にハイパスフィルタ(HPF)
107あるいはローパスフィルタ(LPF)108を通
過し、このようにしてフィルタ107、108を通過し
た信号x1あるいはx2がそれぞれ適応フィルタ104
1、1042に入力されるようになっている。また10
61、1062は分割された周波数帯域の騒音の伝達特
性、1051、1052は各適応フィルタ1041、1
042に対して設けられた係数演算部である。
As one method for improving the problems of this type of Filtered-X algorithm, Japanese Patent Laid-Open No. 28409/1993
The technique of Japanese Patent No. 8 has been proposed. The schematic configuration based on the Filtered-X algorithm of this technology is shown in the figure. In the figure, the same components as those in the above figure are designated by the same reference numerals. )
107 or a low-pass filter (LPF) 108, and the signal x1 or x2 that has passed through the filters 107 and 108 in this manner is applied to the adaptive filter 104, respectively.
1, 1042. Again 10
61 and 1062 are noise transfer characteristics of the divided frequency bands, and 1051 and 1052 are adaptive filters 1041 and 1
This is a coefficient calculator provided for 042.

【0006】この従来例は全帯域に亙って消音効果を得
られる面では優れている。しかしながら前記伝達特性1
061、1062を求めるためのイニシャライズを各フ
ィルタ1041、1042の特性を変える度にやり直す
必要があるという欠点があった。
This conventional example is excellent in that it can obtain a sound deadening effect over the entire band. However, the transfer characteristic 1
There is a drawback that the initialization for obtaining 061 and 1062 needs to be redone each time the characteristics of the filters 1041 and 1042 are changed.

【0007】また消音用スピーカ102と騒音検出用マ
イクロフォン101とが近接している場合に、係数演算
部1051、1052で更新された適応フィルタ104
1、1042の係数は、時間軸で見た場合その遅延部分
を短くするように更新する必要があるが、この時前記フ
ィルタ107、108を信号が通過することによって騒
音検出の時点から適応フィルタ1041、1042へ到
達するまでの時間に遅れが生じ、このフィルタ107、
108の遅延特性によっては係数演算部1051、10
52で更新された適応フィルタ1041、1042の係
数が時間軸で見た場合、騒音の発生に間に合わないとい
う問題点があった。
Further, when the noise reduction speaker 102 and the noise detection microphone 101 are close to each other, the adaptive filter 104 updated by the coefficient calculation units 1051 and 1052.
The coefficients 1 and 1042 need to be updated so as to shorten the delay portion when viewed on the time axis. At this time, the signal passes through the filters 107 and 108, and the adaptive filter 1041 is detected from the time of noise detection. , 1042, there is a delay in reaching the filter 1042.
Depending on the delay characteristics of 108, the coefficient calculators 1051, 10
When the coefficients of the adaptive filters 1041 and 1042 updated in 52 are viewed on the time axis, there is a problem in that noise cannot be generated in time.

【0008】[0008]

【発明が解決しようとする課題】このように従来の方法
では係数演算部に入る信号を手前でフィルタリングする
ことによる遅れのため、適応フィルタの係数の更新が適
時に行えないという問題点があった。
As described above, the conventional method has a problem in that the coefficient of the adaptive filter cannot be updated in a timely manner due to the delay caused by filtering the signal entering the coefficient calculation section in advance. .

【0009】本発明はかかる従来技術の問題点に鑑み、
騒音の周波数のどの帯域においても十分な消音効果を有
したままで適応フィルタの係数を時間軸で見た場合でも
常に騒音の発生に間に合わせることのできる消音装置を
提供することを目的とする。
The present invention has been made in view of the above problems of the prior art.
An object of the present invention is to provide a silencer capable of always keeping up with the generation of noise even when the coefficient of the adaptive filter is viewed on the time axis while having a sufficient noise reduction effect in any frequency band of noise.

【0010】[0010]

【課題を解決するための手段】本発明は、騒音源が発す
る騒音を検出して騒音信号を出力する騒音検出器と、該
騒音源の発する騒音を打ち消す付加音を発する消音用ス
ピーカと、これら騒音源の発する騒音と付加音との合成
音を検出してエラー信号を出力するエラー検出器と、前
記騒音信号を入力とし前記スピーカの駆動信号を作成す
る適応フィルタと、前記騒音信号及びエラー信号に基づ
いて前記適応フィルタの係数を算出する係数演算部と、
前記騒音検出器〜係数演算部間及び前記騒音検出器〜係
数演算部間にそれぞれ設けられた同一特性の帯域通過フ
ィルタとよりなる。
According to the present invention, there is provided a noise detector for detecting a noise emitted from a noise source and outputting a noise signal, and a muffling speaker for emitting an additional sound for canceling the noise emitted by the noise source. An error detector that detects a synthesized sound of noise generated by a noise source and an additional sound and outputs an error signal, an adaptive filter that inputs the noise signal and creates a drive signal for the speaker, the noise signal and the error signal A coefficient calculation unit that calculates the coefficient of the adaptive filter based on
The bandpass filters having the same characteristics are provided between the noise detector and the coefficient calculation unit and between the noise detector and the coefficient calculation unit.

【0011】[0011]

【作用】それぞれの係数演算部の両側に同一特性の帯域
通過フィルタを設けることによって、係数演算部に入る
信号は各帯域通過フィルタの特性を持った信号となるた
め、係数演算部で更新される適応フィルタの係数はそれ
ぞれの帯域通過フィルタの周波数特性を持った係数であ
り、また適応フィルタにかけられる騒音信号は騒音検出
器からの時間遅れのない信号であるため、適応フィルタ
の係数更新に対して時間遅れもなく、また全帯域に亙る
消音が可能となる。
By providing the band pass filters having the same characteristics on both sides of each coefficient calculation section, the signal entering the coefficient calculation section becomes a signal having the characteristics of each band pass filter, and is updated by the coefficient calculation section. The coefficient of the adaptive filter is the coefficient with the frequency characteristics of each band pass filter, and the noise signal applied to the adaptive filter is the signal without time delay from the noise detector. There is no time delay and it is possible to mute sound over the entire band.

【0012】[0012]

【実施例】以下本発明の適応能動消音装置の1実施例に
ついて図面に沿って詳細に説明する。図1は適応能動消
音装置の構成を示すブロック図であり、1は騒音検出器
としての第1マイクロフォン、2は前記第1マイクロフ
ォンによる騒音を打ち消す付加音を発生する消音用スピ
ーカ、3は前記騒音と付加音との合成音を検出するエラ
ー検出器としての第2マイクロフォン、H2は前記消音
用スピーカ2から出力された付加音信号が第2マイクロ
フォン3に入るまでの伝達特性、H1は畳み込み用フィ
ルタ、ADFi (i=1〜n)は前記第1マイクロフォ
ン1の検出信号に基づいて前記消音用スピーカ2に駆動
信号を供給する適応フィルタ、LMS i (i=1〜n)
は前記各適応フィルタADFi のそれぞれの係数を更新
するための係数演算部、FILi は前記第1マイクロフ
ォン1〜係数演算部LMSi 間及び第2マイクロフォン
3〜係数演算部LMSi 間にそれぞれ同一特性のものを
一対ずつ挿入された帯域通過フィルタである。
DETAILED DESCRIPTION OF THE PREFERRED EMBODIMENTS One embodiment of the adaptive active noise reduction system of the invention will be described below.
The details will be described with reference to the drawings. Figure 1 shows adaptive active erase
1 is a block diagram showing a configuration of a sound device, and 1 is a noise detector
As a first microphone, 2 is the first microphone
Sound-suppressing sound that generates additional sound that cancels the noise caused by
An encoder 3 is an error detecting a synthesized sound of the noise and the additional sound.
-The second microphone as a detector, H2 is the mute
The additional sound signal output from the speaker 2 for the second microphone
Transfer characteristics before entering phone 3, H1 is the convolution filter
Ruta, ADFi (I = 1 to n) is the first microphone
Driven to the muffling speaker 2 based on the detection signal of
Adaptive filter for supplying signal, LMS i (I = 1 to n)
Is each adaptive filter ADFi Update each coefficient of
Coefficient calculation unit for performing FILi Is the first microf
ON 1-coefficient calculation unit LMSi Between and second microphone
3 to coefficient calculation unit LMSi Have the same characteristics between
It is a bandpass filter inserted in pairs.

【0013】なお、ここでは係数演算部LMSi として
従来周知のLMS(Least Mean Square )アルゴリズム
を用い、帯域通過フィルタFILi として従来周知のロ
ーパスフィルタを用いた。
Here, a well-known LMS (Least Mean Square) algorithm is used as the coefficient calculation unit LMS i , and a well-known low-pass filter is used as the band-pass filter FIL i .

【0014】かかる構成において、第1マイクロフォン
1にて検出した騒音信号xに適応フィルタADFi を畳
み込んでスピーカ駆動信号を作り、消音用スピーカ2を
駆動して付加音を出力する。一方前記信号xにフィルタ
H1を畳み込み、さらに帯域通過フィルタFILi を畳
み込んだ信号をzとする。また前記第2マイクロフォン
3からのエラー信号eに帯域フィルタFILi を畳み込
んだ信号をfとする。
In such a configuration, the noise signal x detected by the first microphone 1 is convoluted with the adaptive filter ADF i to generate a speaker drive signal, and the muffling speaker 2 is driven to output an additional sound. On the other hand, a signal obtained by convolving the signal x with the filter H1 and further convolving the band pass filter FIL i is represented by z. Further, a signal obtained by convolving the band-pass filter FIL i with the error signal e from the second microphone 3 is defined as f.

【0015】こうして得られた信号z、fを係数演算部
LMSi に入力して適応フィルタADFi の係数更新を
LMSアルゴリズムを用いて行う。また第1マイクロフ
ォン1への消音用スピーカ2からのフィードバック音の
対策として、消音用スピーカ2から出た付加音信号が第
1マイクロフォン1に入るまでの伝達特性H2をキャン
セルさせるように構成した。
The signals z and f thus obtained are input to the coefficient calculator LMS i to update the coefficients of the adaptive filter ADF i using the LMS algorithm. Further, as a measure against the feedback sound from the muffling speaker 2 to the first microphone 1, the transfer characteristic H2 until the additional sound signal emitted from the muffling speaker 2 enters the first microphone 1 is canceled.

【0016】本実施例の性能を従来の消音装置と比較す
るため、本実施例を図2に示すようなひとつの適応フィ
ルタADFを持つ仕様とし、比較例として従来技術のと
ころで説明した特開平3−284098号公報の技術を
同様に図3に示すような1個の適応フィルタADFを持
つ仕様として、サンプリング周波数5500Hz、図示
しないA/D、D/A変換器のローパスフィルタのカッ
トオフ周波数を2000Hz、フィルタH1、伝達特性
H2、適応フィルタADFのフィルタタップ数を200
に設定し、帯域通過FILの周波数特性を図4に示すよ
うな特性とし、且つ帯域通過FILの遅延時間を1.9
5msecに設定してそれぞれの消音装置について比較
実験を行った。
In order to compare the performance of this embodiment with a conventional silencer, this embodiment has a specification having one adaptive filter ADF as shown in FIG. Similarly, the technique of Japanese Patent No. 284098 has a specification of having one adaptive filter ADF as shown in FIG. 3, with a sampling frequency of 5500 Hz and a cutoff frequency of a low-pass filter of an A / D and D / A converter (not shown) of 2000 Hz. , The filter H1, the transfer characteristic H2, and the number of filter taps of the adaptive filter ADF are 200
, The frequency characteristic of the band-pass FIL is set as shown in FIG. 4, and the delay time of the band-pass FIL is 1.9.
A comparative experiment was conducted for each muffler set to 5 msec.

【0017】図5は上記比較実験の結果得られた第1マ
イクロフォン1(101)と消去用スピーカ2(10
2)との距離と消音量との関係を示すものである。この
結果本実施例(図中実線で示す)の場合、第1マイクロ
フォン1と消音用スピーカ2を距離に関係なく比較的大
きな消音効果が得られているのに対し、比較例(図中点
線で示す)では約70cmの距離までは直線的に一定の
勾配もって消音効果が減少しそれより近づくとほとんど
消音効果は変動せず小さいままであることがわかる。こ
の距離70cmは帯域通過フィルタFILの遅延時間
1.95msecを距離に換算した値である66.3c
mにほぼ一致しており、比較例の消音装置の場合は帯域
通過フィルタとして用いるフィルタの遅延特性の影響を
受けたためである。これに対して本実施例では帯域通過
フィルタFILの遅延特性に全く影響されることなく全
周波数帯域の消音が可能になるといえる。
FIG. 5 shows a first microphone 1 (101) and an erasing speaker 2 (10) obtained as a result of the above comparative experiment.
2) shows the relationship between the distance to 2) and the volume of noise. As a result, in the case of the present embodiment (shown by the solid line in the figure), a relatively large silencing effect is obtained regardless of the distance between the first microphone 1 and the muffling speaker 2, whereas the comparative example (indicated by the dotted line in the figure). It can be seen that the noise reduction effect decreases linearly with a constant slope up to a distance of about 70 cm, and the noise reduction effect does not fluctuate and remains small when the distance approaches 70 cm. This distance 70 cm is a value obtained by converting the delay time of the band pass filter FIL of 1.95 msec into a distance of 66.3 c.
This is because it substantially coincides with m, and in the case of the silencer of the comparative example, it is affected by the delay characteristic of the filter used as the bandpass filter. On the other hand, in this embodiment, it can be said that the muffling of the entire frequency band is possible without being affected by the delay characteristics of the band pass filter FIL.

【0018】[0018]

【発明の効果】本発明は以上の説明のごとく、帯域通過
フィルタの遅延特性による影響をまったく受けないた
め、騒音検出器と消音用スピーカを近づけても常に適応
フィルタの係数の更新が適時に行え、全周波数帯域に亙
って騒音を消去できる効果がある。
As described above, the present invention is not affected by the delay characteristics of the bandpass filter at all, and therefore the coefficient of the adaptive filter can be updated at a proper time even if the noise detector and the muffling speaker are brought close to each other. , Has the effect of eliminating noise over the entire frequency band.

【図面の簡単な説明】[Brief description of drawings]

【図1】本発明の適応能動消音装置の一実施例を示すブ
ロック図である。
FIG. 1 is a block diagram showing an embodiment of an adaptive active silencer of the present invention.

【図2】実験に用いた図1の装置を簡易化した消音装置
の実施例のブロック図である。
FIG. 2 is a block diagram of an embodiment of a silencer that is a simplified version of the device of FIG. 1 used in an experiment.

【図3】実験に用いた簡易化された比較例の消音装置の
ブロック図である。
FIG. 3 is a block diagram of a simplified silencer of a comparative example used in an experiment.

【図4】図2及び図3の帯域通過フィルタの周波数特性
を示す図である。
FIG. 4 is a diagram showing frequency characteristics of the bandpass filters of FIGS. 2 and 3;

【図5】実施例と比較例との消音効果の実験結果を示す
図である。
FIG. 5 is a diagram showing an experimental result of a sound deadening effect between an example and a comparative example.

【図6】Filtered-Xアルゴリズムの例を示すブロック図
である。
FIG. 6 is a block diagram showing an example of a Filtered-X algorithm.

【図7】図1に相当する従来の消音装置の構成を示すブ
ロック図である。
FIG. 7 is a block diagram showing a configuration of a conventional silencer corresponding to FIG.

【符号の説明】[Explanation of symbols]

1 騒音検出器 2 消去用スピーカ 3 エラー検出器 ADFi 適応フィルタ LMSi 係数演算部 FILi 帯域通過フィルタ1 Noise Detector 2 Elimination Speaker 3 Error Detector ADF i Adaptive Filter LMS i Coefficient Calculator FIL i Bandpass Filter

Claims (1)

【特許請求の範囲】[Claims] 【請求項1】 騒音源が発する騒音を検出して騒音信号
を出力する騒音検出器と、該騒音源の発する騒音を打ち
消す付加音を発する消音用スピーカと、これら騒音源の
発する騒音と付加音との合成音を検出してエラー信号を
出力するエラー検出器と、前記騒音信号を入力とし前記
スピーカの駆動信号を作成する適応フィルタと、前記騒
音信号及びエラー信号に基づいて前記適応フィルタの係
数を算出する係数演算部と、前記騒音検出器〜係数演算
部間及び前記騒音検出器〜係数演算部間にそれぞれ設け
られた同一特性の帯域通過フィルタとよりなる適応能動
消音装置。
1. A noise detector that detects noise emitted from a noise source and outputs a noise signal, a noise canceling speaker that emits an additional sound to cancel the noise emitted from the noise source, and a noise and an additional sound emitted from these noise sources. An error detector for detecting a synthesized sound of and outputting an error signal, an adaptive filter for inputting the noise signal to create a drive signal for the speaker, and a coefficient of the adaptive filter based on the noise signal and the error signal An adaptive active silencer comprising: a coefficient calculating unit for calculating the above; and a band pass filter having the same characteristic, which is provided between the noise detector and the coefficient calculating unit and between the noise detector and the coefficient calculating unit.
JP4281588A 1992-10-20 1992-10-20 Adaptive active muffling device Pending JPH06130968A (en)

Priority Applications (1)

Application Number Priority Date Filing Date Title
JP4281588A JPH06130968A (en) 1992-10-20 1992-10-20 Adaptive active muffling device

Applications Claiming Priority (1)

Application Number Priority Date Filing Date Title
JP4281588A JPH06130968A (en) 1992-10-20 1992-10-20 Adaptive active muffling device

Publications (1)

Publication Number Publication Date
JPH06130968A true JPH06130968A (en) 1994-05-13

Family

ID=17641247

Family Applications (1)

Application Number Title Priority Date Filing Date
JP4281588A Pending JPH06130968A (en) 1992-10-20 1992-10-20 Adaptive active muffling device

Country Status (1)

Country Link
JP (1) JPH06130968A (en)

Cited By (5)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
JPH0777995A (en) * 1993-04-27 1995-03-20 Hughes Aircraft Co Active noise-erasing device
JPH07334170A (en) * 1994-06-07 1995-12-22 Daidan Kk Speaker device for active noise control system
JPH07334167A (en) * 1994-06-07 1995-12-22 Daidan Kk Sensor device for active noise control system
JP2010097018A (en) * 2008-10-17 2010-04-30 Nippon Sharyo Seizo Kaisha Ltd Adapted control system
US9392380B2 (en) 2014-07-07 2016-07-12 Rion Co., Ltd. Hearing aid and feedback canceller

Cited By (5)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
JPH0777995A (en) * 1993-04-27 1995-03-20 Hughes Aircraft Co Active noise-erasing device
JPH07334170A (en) * 1994-06-07 1995-12-22 Daidan Kk Speaker device for active noise control system
JPH07334167A (en) * 1994-06-07 1995-12-22 Daidan Kk Sensor device for active noise control system
JP2010097018A (en) * 2008-10-17 2010-04-30 Nippon Sharyo Seizo Kaisha Ltd Adapted control system
US9392380B2 (en) 2014-07-07 2016-07-12 Rion Co., Ltd. Hearing aid and feedback canceller

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