JPH07325588A - Muffler - Google Patents
MufflerInfo
- Publication number
- JPH07325588A JPH07325588A JP6121171A JP12117194A JPH07325588A JP H07325588 A JPH07325588 A JP H07325588A JP 6121171 A JP6121171 A JP 6121171A JP 12117194 A JP12117194 A JP 12117194A JP H07325588 A JPH07325588 A JP H07325588A
- Authority
- JP
- Japan
- Prior art keywords
- noise
- output
- adaptive filter
- microphone
- coefficient
- 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
Links
Landscapes
- Duct Arrangements (AREA)
- Soundproofing, Sound Blocking, And Sound Damping (AREA)
- Measurement Of Mechanical Vibrations Or Ultrasonic Waves (AREA)
Abstract
Description
【０００１】[0001]
【産業上の利用分野】本発明は能動的騒音制御を用いた
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 a muffler using active noise control.
【０００２】[0002]
【従来の技術】近年、特開平４−２８２６９４号公報、
特開平４−２８２６９５号公報、特開平４−２８２６９
６号公報などに見られるように、送風機を有する空調機
などの騒音をディジタル信号処理技術を用いてスピーカ
から制御音を出力し、送風ダクト内で消音する能動的騒
音制御が提案されている。2. Description of the Related Art Recently, Japanese Unexamined Patent Publication No. 4-282694,
JP-A-4-282695 and JP-A-4-28269
As disclosed in Japanese Patent No. 6 or the like, active noise control has been proposed in which noise of an air conditioner having a blower is output from a speaker by using a digital signal processing technique and the noise is silenced in a blower duct.
【０００３】以下、その構成について図５を参照しなが
ら説明する。図に示すように、消音装置は送風ダクト８
内に、第１の騒音検出器であるマイクロホン１ａと第２
の騒音検出器であるマイクロホン１ｂと、スピーカ７を
配し、制御装置１００内に制御回路１０１と、マイクロ
ホン１ｂの出力をある規定値と比較する比較回路１０２
と、アダプティブフィルタ２の係数更新を停止する停止
回路１０３を配している。The structure will be described below with reference to FIG. As shown in the figure, the muffler includes a ventilation duct 8
The first noise detector includes the microphone 1a and the second noise detector.
The microphone 1b which is the noise detector of the device and the speaker 7 are arranged, and the control circuit 101 in the control device 100 and the comparison circuit 102 which compares the output of the microphone 1b with a predetermined value.
And a stop circuit 103 for stopping the coefficient update of the adaptive filter 2.
【０００４】上記構成において、騒音は送風ダクト８内
を矢印９の方向に伝搬され、マイクロホン１ａで検出さ
れる。その検出信号は制御回路１０１に入力される。同
時にマイクロホン１ｂで検出された誤差信号も制御回路
１０１に入力される。そして制御回路１０１はこの２つ
の信号から、マイクロホン１ｂにおける騒音が減衰する
ようにマイクロホン１ｂでの騒音と同振幅逆位相の制御
音をスピーカ７より再生する。In the above structure, noise is propagated in the air duct 8 in the direction of the arrow 9 and detected by the microphone 1a. The detection signal is input to the control circuit 101. At the same time, the error signal detected by the microphone 1b is also input to the control circuit 101. Then, the control circuit 101 reproduces the control sound having the same amplitude and opposite phase as the noise in the microphone 1b from the speaker 7 so that the noise in the microphone 1b is attenuated from these two signals.
【０００５】以上のようにマイクロホン１ｂからの誤差
信号を常に監視しながらアダプティブフィルタ２の係数
が更新され続けることによって消音していくことになる
が、なんらかの要因により間違った特性になることがあ
る。するとマイクロホン１ｂで騒音付加となることがあ
るので、比較回路１０２に入力された誤差信号レベルを
ある規定値と比較し、規定値を越えた場合は停止回路１
０３によって制御回路１０１の動作を停止することとな
る。As described above, while the error signal from the microphone 1b is always monitored, the coefficient of the adaptive filter 2 is continuously updated to mute the sound. However, the characteristic may be incorrect due to some factor. Then, noise may be added to the microphone 1b. Therefore, the error signal level input to the comparison circuit 102 is compared with a certain specified value, and if it exceeds the specified value, the stop circuit 1
By 03, the operation of the control circuit 101 is stopped.
【０００６】[0006]
【発明が解決しようとする課題】このような従来の図５
に示すようなマイクロホン１ｂの誤差信号レベルを規定
値と比較して係数更新を停止する構成では、規定値の決
定が非常に困難であった。つまり誤差信号の値は騒音の
大きさ、消音量により決定されるため送風ダクト８が設
置される環境により異なることとなる、そのためこの誤
差信号と比較する規定値も同様に環境によって異なるこ
ととなる。また騒音付加が発生してから消音装置を停止
するため、使用者も騒音付加を感知することとなり、使
用者に対して不快感を与えるという問題があった。FIG. 5 of the related art described above.
In a configuration in which the error signal level of the microphone 1b is compared with the specified value and the coefficient update is stopped, it is very difficult to determine the specified value. In other words, the value of the error signal is determined by the noise level and the volume of noise, so that it varies depending on the environment in which the air duct 8 is installed. Therefore, the specified value to be compared with this error signal also varies depending on the environment. . Further, since the muffling device is stopped after the noise is added, the user senses the addition of the noise, which causes a problem that the user feels uncomfortable.
【０００７】本発明は上記課題を解決するもので、送風
ダクト８が設置される環境により規定値を変更すること
なく、十分の消音量が得られた時点で係数更新を停止す
ることによって安全に騒音制御できる消音装置を提供す
ることにある。The present invention solves the above-mentioned problems, and the coefficient update is stopped safely when a sufficient volume is obtained without changing the specified value depending on the environment in which the air duct 8 is installed. An object of the present invention is to provide a silencer capable of controlling noise.
【０００８】第２の目的は検出される騒音の周波数特性
と、スピーカ７からマイクロホン１ｂまでの伝達関数の
関係で十分の消音量を得られない場合にそれを改善し、
十分の消音量が得られた時点で係数更新を停止すること
によって安全に騒音制御できる消音装置を提供すること
にある。The second purpose is to improve the noise characteristic detected and the transfer function from the loudspeaker 7 to the microphone 1b when a sufficient volume cannot be obtained.
An object of the present invention is to provide a silencer capable of safely controlling noise by stopping the updating of coefficients when a sufficient silence level is obtained.
【０００９】第３の目的は検出される騒音の周波数特性
と、スピーカ７からマイクロホン１ｂまでの伝達関数の
関係で十分の消音量を得られない場合にそれを改善し、
十分の消音量が得られた時点で係数更新を停止すること
によって安全に騒音制御できる消音装置を提供すること
にある。A third object is to improve the noise characteristic detected and the transfer function from the loudspeaker 7 to the microphone 1b when a sufficient volume cannot be obtained.
An object of the present invention is to provide a silencer capable of safely controlling noise by stopping the updating of coefficients when a sufficient silence level is obtained.
【００１０】[0010]
【課題を解決するための手段】本発明の消音装置は上記
第１の目的を達成するために、第１の手段は騒音源を伝
搬する騒音を検出する第１の騒音検出機と、前記第１の
騒音検出器の出力を信号処理する演算器と、前記演算器
の出力を適応制御するアダプティブフィルタと、前記演
算器の出力を信号処理する第１のディジタルフィルタ
と、前記第１の騒音検出器の設置位置に比べて騒音源か
ら遠い位置に設置され、前記アダプティブフィルタの出
力を再生するスピーカと、前記アダプティブフィルタの
出力を信号処理してその結果を前記演算器に入力する第
２のディジタルフィルタと、前記スピーカの近傍の騒音
源から遠い位置に設置された第２の騒音検出器と、前記
第２の騒音検出器の出力と前記ディジタルフィルタの出
力から前記アダプティブフィルタの時間領域における係
数を演算して更新する係数演算器と、前記第２の騒音検
出器の出力を演算して時間領域における平均値を算出す
る平均化手段と、前記平均化手段の出力を単位時間毎に
比較し前記出力が収束したことを検出する比較手段と、
前記比較手段の出力から前記係数演算器による係数更新
を停止する停止手段の構成とする。In order to achieve the above first object, the silencer of the present invention has a first means for detecting noise propagating through a noise source, and No. 1 noise detector output signal processing, an adaptive filter adaptively controlling the output of the arithmetic unit, a first digital filter signal processing the output of the arithmetic unit, the first noise detection A speaker installed at a position farther from the noise source than the installation position of the device and reproducing the output of the adaptive filter, and a second digital device for processing the output of the adaptive filter and inputting the result to the arithmetic unit. A filter, a second noise detector installed at a position far from a noise source near the speaker, an output of the second noise detector and an output of the digital filter, and the adaptor. A coefficient calculator for calculating and updating a coefficient in the time domain of the filter, an averaging means for calculating an output of the second noise detector to calculate an average value in the time domain, and an output of the averaging means. Comparing means for detecting that the output has converged by comparing every unit time,
The stop means is configured to stop the coefficient update by the coefficient calculator from the output of the comparison means.
【００１１】また、第２の目的を達成するために、第２
の手段は騒音源を伝搬する騒音を検出する第１の騒音検
出機と、前記第１の騒音検出器の出力を信号処理する演
算器と、前記演算器の出力を適応制御するアダプティブ
フィルタと、前記演算器の出力を信号処理する第１のデ
ィジタルフィルタと、前記アダプティブフィルタの出力
を信号処理してその結果を前記演算器に入力する第２の
ディジタルフィルタと、前記アダプティブフィルタの出
力の周波数特性を調整する信号処理回路と、前記第１の
騒音検出器の設置位置に比べて騒音源から遠い位置に設
置され、前記信号処理回路の出力を再生するスピーカ
と、前記スピーカの近傍の騒音源から遠い位置に設置さ
れた第２の騒音検出器と、前記信号処理回路の出力と前
記ディジタルフィルタの出力から前記アダプティブフィ
ルタの時間領域における係数を演算して更新する係数演
算器と、前記第２の騒音検出器の出力を演算して時間領
域における平均値を算出する平均化手段と、前記平均化
手段の出力を単位時間毎に比較し前記出力が収束したこ
とを検出する比較手段と、前記比較手段の出力から前記
係数演算器による係数更新を停止する停止手段の構成と
する。In order to achieve the second object, the second
Means for detecting noise propagating through a noise source, a calculator for signal processing the output of the first noise detector, and an adaptive filter for adaptively controlling the output of the calculator. A first digital filter for signal processing the output of the arithmetic unit, a second digital filter for signal processing the output of the adaptive filter and inputting the result to the arithmetic unit, and frequency characteristics of the output of the adaptive filter. A signal processing circuit for adjusting the signal, a speaker installed farther from the noise source than the installation position of the first noise detector, and reproducing the output of the signal processing circuit, and a noise source near the speaker. The second noise detector installed at a distant position, the output of the signal processing circuit and the output of the digital filter are placed in the time domain of the adaptive filter. A coefficient calculator for calculating and updating a coefficient, an averaging means for calculating an output of the second noise detector to calculate an average value in a time domain, and an output of the averaging means for each unit time. The comparison means is configured to compare and detect that the output has converged, and the stop means for stopping the coefficient update by the coefficient calculator from the output of the comparison means.
【００１２】また、第３の目的を達成するために、第３
の手段は騒音源を伝搬する騒音を検出する第１の騒音検
出機と、前記第１の騒音検出器の出力を信号処理する演
算器と、前記演算器の出力を適応制御するアダプティブ
フィルタと、前記演算器の出力を信号処理する第１のデ
ィジタルフィルタと、前記第１の騒音検出器の設置位置
に比べて騒音源から遠い位置に設置され、前記アダプテ
ィブフィルタの出力を再生するスピーカと、前記アダプ
ティブフィルタの出力を信号処理してその結果を前記演
算器に入力する第２のディジタルフィルタと、前記スピ
ーカの近傍の騒音源から遠い位置に設置された第２の騒
音検出器と、前記第２の騒音検出器の出力の周波数特性
を調整する信号処理回路と、前記信号処理回路の出力と
前記ディジタルフィルタの出力から前記アダプティブフ
ィルタの時間領域における係数を演算して更新する係数
演算器と、前記第２の騒音検出器の出力を演算して時間
領域における平均値を算出する平均化手段と、前記平均
化手段の出力を単位時間毎に比較し前記出力が収束した
ことを検出する比較手段と、前記比較手段の出力から前
記係数演算器による係数更新を停止する停止手段の構成
とする。In order to achieve the third object, the third
Means for detecting noise propagating through a noise source, a calculator for signal processing the output of the first noise detector, and an adaptive filter for adaptively controlling the output of the calculator. A first digital filter that performs signal processing on the output of the arithmetic unit; a speaker that is installed farther from a noise source than the installation position of the first noise detector, and that reproduces the output of the adaptive filter; A second digital filter for processing the output of the adaptive filter and inputting the result to the arithmetic unit; a second noise detector installed at a position far from a noise source near the speaker; Signal processing circuit for adjusting the frequency characteristic of the output of the noise detector, and the time domain of the adaptive filter from the output of the signal processing circuit and the output of the digital filter A coefficient calculator for calculating and updating a coefficient in the unit, an averaging means for calculating an output of the second noise detector to calculate an average value in a time domain, and an output of the averaging means for each unit time. The comparison means is configured to compare and detect that the output has converged, and the stop means for stopping the coefficient update by the coefficient calculator from the output of the comparison means.
【００１３】[0013]
【作用】本発明は上記した第１の手段の構成により、第
２の騒音検出器の検出信号を監視することにより、アダ
プティブフィルタの係数更新が収束することを検出し、
騒音付加となる前にアダプティブフィルタの係数の更新
を止めることができ、発散することなく安全に騒音制御
ができるものである。According to the present invention, the configuration of the above-described first means monitors the detection signal of the second noise detector to detect that the coefficient update of the adaptive filter is converged,
The updating of the coefficient of the adaptive filter can be stopped before noise is added, and noise can be safely controlled without diverging.
【００１４】また、第２の手段の構成によりアダプティ
ブフィルタの出力の周波数特性を調整することで十分の
消音量を得、第２の騒音検出器の検出信号を監視するこ
とにより、アダプティブフィルタの係数更新が収束する
ことを検出し、騒音付加となる前にアダプティブフィル
タの係数の更新を止めることができ、発散することなく
安全に騒音制御ができるものである。Further, by adjusting the frequency characteristic of the output of the adaptive filter by the constitution of the second means, a sufficient sound volume is obtained, and by monitoring the detection signal of the second noise detector, the coefficient of the adaptive filter is adjusted. It is possible to detect the convergence of the update and stop the update of the coefficient of the adaptive filter before the noise is added, so that the noise can be safely controlled without diverging.
【００１５】また、第３の手段の構成により第２の騒音
検出器の出力の周波数特性を調整することで十分の消音
量を得、第２の騒音検出器の検出信号を監視することに
より、アダプティブフィルタの係数更新が収束すること
を検出し、騒音付加となる前にアダプティブフィルタの
係数の更新を止めることができ、発散することなく安全
に騒音制御ができるものである。In addition, by adjusting the frequency characteristic of the output of the second noise detector by the configuration of the third means, a sufficient noise reduction level is obtained, and the detection signal of the second noise detector is monitored. It is possible to detect the convergence of the update of the coefficient of the adaptive filter, stop the update of the coefficient of the adaptive filter before noise is added, and safely perform noise control without divergence.
【００１６】[0016]
【実施例】以下、本発明の第１実施例について図１およ
び図２を参照しながら説明する。DETAILED DESCRIPTION OF THE PREFERRED EMBODIMENTS A first embodiment of the present invention will be described below with reference to FIGS.
【００１７】図１に示すように、消音装置は送風ダクト
８内に、第１の騒音検出器であるマイクロホン１ａと第
２の騒音検出器であるマイクロホン１ｂと、スピーカ７
を配し、制御装置２０内に演算器６と、アダプティブフ
ィルタ２と、ＦＩＲフィルタ５と、ＦＩＲフィルタ３
と、アダプティブフィルタ２の係数更新を実現するＬＭ
Ｓ演算器４と、マイクロホン１ｂの出力を平均化する平
均化手段１０と、この平均化手段の出力を単位時間毎に
比較する比較手段１１と、アダプティブフィルタ２の係
数更新を停止する停止手段１２を配している。As shown in FIG. 1, the muffler has a microphone 1a as a first noise detector, a microphone 1b as a second noise detector, and a speaker 7 in a blower duct 8.
And an arithmetic unit 6, an adaptive filter 2, an FIR filter 5, and an FIR filter 3 in the control device 20.
And an LM that realizes coefficient updating of the adaptive filter 2.
S calculator 4, averaging means 10 for averaging the output of the microphone 1b, comparing means 11 for comparing the output of this averaging means for each unit time, and stopping means 12 for stopping the coefficient update of the adaptive filter 2. Are arranged.
【００１８】上記構成において、騒音は送風ダクト８内
を矢印９の方向に伝搬され、マイクロホン１ａで検出さ
れる。その検出信号と音響フィードバック補正用のＦＩ
Ｒフィルタ５の出力信号は演算器６に入力され、ＦＩＲ
フィルタ５の出力信号はマイクロホン１ａの検出信号に
対して逆位相で加算される。ここでＦＩＲフィルタ５に
は予めスピーカ７からマイクロホン１ａまでの伝達関数
が係数として近似されている。演算器６の出力はアダプ
ティブフィルタ２とＦＩＲフィルタ３に入力され、アダ
プティブフィルタ２で信号処理された信号はＦＩＲフィ
ルタ５とスピーカ７に入力される。このスピーカ７で再
生された制御音とダクト内を伝搬されてくる騒音は、マ
イクロホン１ｂの位置で音響的に加算され、誤差信号と
してＬＭＳ演算器４と平均化手段１０に入力される。Ｌ
ＭＳ演算器４はこの誤差信号とＦＩＲフィルタ３からの
出力信号により、アダプティブフィルタ２の係数を演算
し、マイクロホン１ｂにおいて騒音が減衰するように係
数更新を行う。ここでＦＩＲフィルタ３には予めスピー
カ７からマイクロホン１ｂまでの伝達関数が係数として
近似されている。このようにマイクロホン１ｂからの誤
差信号を常に監視しながらアダプティブフィルタ２の係
数が更新され続けることによって消音していくことにな
るが、なんらかの要因により間違った特性になることが
ある。するとマイクロホン１ｂで騒音付加となることが
ある。そこで平均化手段１０では時間変動を伴うマイク
ロホン１ｂの誤差信号を時間平均し、その出力を比較手
段１１に入力するものとする。比較手段１１では図２に
示すように平均化された誤差信号を単位時間毎に比較し
て、誤差信号が収束したかどうかを判定する、そこで誤
差信号が収束したと判定されると、停止手段１２により
ＬＭＳ演算器４によるアダプティブフィルタ２の係数更
新を停止することができる。In the above structure, noise is propagated in the air duct 8 in the direction of the arrow 9 and detected by the microphone 1a. FI for the detection signal and acoustic feedback correction
The output signal of the R filter 5 is input to the computing unit 6 and the FIR
The output signal of the filter 5 is added in the opposite phase to the detection signal of the microphone 1a. Here, the transfer function from the speaker 7 to the microphone 1a is approximated to the FIR filter 5 in advance as a coefficient. The output of the computing unit 6 is input to the adaptive filter 2 and the FIR filter 3, and the signal processed by the adaptive filter 2 is input to the FIR filter 5 and the speaker 7. The control sound reproduced by the speaker 7 and the noise propagating in the duct are acoustically added at the position of the microphone 1b and input to the LMS calculator 4 and the averaging means 10 as an error signal. L
The MS calculator 4 calculates the coefficient of the adaptive filter 2 based on this error signal and the output signal from the FIR filter 3, and updates the coefficient so that noise is attenuated in the microphone 1b. Here, the transfer function from the speaker 7 to the microphone 1b is approximated to the FIR filter 3 in advance as a coefficient. While the error signal from the microphone 1b is constantly monitored in this way, the coefficient of the adaptive filter 2 continues to be updated, so that the sound is muted, but due to some factor, the characteristic may be incorrect. Then, noise may be added to the microphone 1b. Therefore, it is assumed that the averaging means 10 time-averages the error signal of the microphone 1b that is subject to time fluctuation, and the output thereof is input to the comparing means 11. The comparing means 11 compares the averaged error signals as shown in FIG. 2 every unit time to determine whether the error signal has converged. When it is determined that the error signal has converged, the stopping means 12 allows the coefficient update of the adaptive filter 2 by the LMS calculator 4 to be stopped.
【００１９】このように本発明の第１実施例の消音装置
によれば、誤差信号を監視することにより、アダプティ
ブフィルタの係数更新が収束することを検出し、騒音付
加となる前にアダプティブフィルタの係数の更新を止め
ることができ、発散することなく安全に騒音制御ができ
るものである。As described above, according to the silencer of the first embodiment of the present invention, by monitoring the error signal, it is detected that the coefficient update of the adaptive filter is converged, and the adaptive filter of the adaptive filter is detected before noise is added. The coefficient update can be stopped, and noise can be safely controlled without diverging.
【００２０】つぎに本発明の第２の実施例について図３
を参照しながら説明する。図３に示すように、消音装置
は送風ダクト８内に、第１の騒音検出器であるマイクロ
ホン１ａと第２の騒音検出器であるマイクロホン１ｂ
と、スピーカ７を配し、制御装置２０内に演算器６と、
アダプティブフィルタ２と、信号処理回路であるところ
のイコライザ１３と、ＦＩＲフィルタ５と、ＦＩＲフィ
ルタ３と、アダプティブフィルタ２の係数更新を実現す
るＬＭＳ演算器４と、マイクロホン１ｂの出力を平均化
する平均化手段１０と、この平均化手段の出力を単位時
間毎に比較する比較手段１１と、アダプティブフィルタ
２の係数更新を停止する停止手段１２を配している。Next, a second embodiment of the present invention will be described with reference to FIG.
Will be described with reference to. As shown in FIG. 3, the muffler has a microphone 1a as a first noise detector and a microphone 1b as a second noise detector in the air duct 8.
And a speaker 7, and a computing unit 6 in the control device 20,
Adaptive filter 2, equalizer 13 that is a signal processing circuit, FIR filter 5, FIR filter 3, LMS calculator 4 that implements coefficient updating of adaptive filter 2, and average that averages the outputs of microphone 1b. Arrangement means 10, comparison means 11 for comparing the output of the averaging means for each unit time, and stop means 12 for stopping the coefficient update of the adaptive filter 2 are arranged.
【００２１】上記構成において、騒音は送風ダクト８内
を矢印９の方向に伝搬され、マイクロホン１ａで検出さ
れる。その検出信号と音響フィードバック補正用のＦＩ
Ｒフィルタ５の出力信号は演算器６に入力され、ＦＩＲ
フィルタ５の出力信号はマイクロホン１ａの検出信号に
対して逆位相で加算される。ここでＦＩＲフィルタ５に
は予めスピーカ７からマイクロホン１ａまでの伝達関数
が係数として近似されている。演算器６の出力はアダプ
ティブフィルタ２とＦＩＲフィルタ３に入力され、アダ
プティブフィルタ２で信号処理された信号はＦＩＲフィ
ルタ５とイコライザ１３に入力され、イコライザ１３に
よって周波数特性を調整されてスピーカ７より再生され
る。このスピーカ７で再生された制御音とダクト内を伝
搬されてくる騒音は、マイクロホン１ｂの位置で音響的
に加算され、誤差信号としてＬＭＳ演算器４と平均化手
段１０に入力される。ＬＭＳ演算器４はこの誤差信号と
ＦＩＲフィルタ３からの出力信号により、アダプティブ
フィルタ２の係数を演算し、マイクロホン１ｂにおいて
騒音が減衰するように係数更新を行う。ここでＦＩＲフ
ィルタ３には予めスピーカ７からマイクロホン１ｂまで
の伝達関数が係数として近似されている。In the above structure, noise is propagated in the air duct 8 in the direction of the arrow 9 and detected by the microphone 1a. FI for the detection signal and acoustic feedback correction
The output signal of the R filter 5 is input to the computing unit 6 and the FIR
The output signal of the filter 5 is added in the opposite phase to the detection signal of the microphone 1a. Here, the transfer function from the speaker 7 to the microphone 1a is approximated to the FIR filter 5 in advance as a coefficient. The output of the arithmetic unit 6 is input to the adaptive filter 2 and the FIR filter 3, the signal processed by the adaptive filter 2 is input to the FIR filter 5 and the equalizer 13, and the frequency characteristic is adjusted by the equalizer 13 and reproduced by the speaker 7. To be done. The control sound reproduced by the speaker 7 and the noise propagating in the duct are acoustically added at the position of the microphone 1b and input to the LMS calculator 4 and the averaging means 10 as an error signal. The LMS calculator 4 calculates the coefficient of the adaptive filter 2 based on this error signal and the output signal from the FIR filter 3, and updates the coefficient so that noise is attenuated in the microphone 1b. Here, the transfer function from the speaker 7 to the microphone 1b is approximated to the FIR filter 3 in advance as a coefficient.
【００２２】ここでマイクロホン１ａ、１ｂで検出され
る騒音信号、誤差信号の周波数特性が低域では大きく、
高域では小さい特性を有し、スピーカ７からマイクロホ
ン１ｂまでの伝達関数の周波数特性が低域、中域に比べ
高域が落ち込んでいる特性を有している時、マイクロホ
ン１ｂにおける消音効果は低域になるほど大きな消音量
が得られるが、高域では消音量が小さいあるいは消音で
きない場合がある。このようなとき高域の消音量も低域
と同程度確保したい場合は、図３のようにイコライザ１
３を用い、その周波数特性を高域ほどレベルが大きくな
るようにすれば前記スピーカ７からマイクロホン１ｂま
での伝達関数の周波数特性は、高域の落ち込みが改善さ
れることになるので、マイクロホン１ｂでの消音効果は
各帯域で一様に得られる様になる。Here, the frequency characteristics of the noise signal and the error signal detected by the microphones 1a and 1b are large in the low range,
When the high frequency range has a small characteristic, and the frequency characteristic of the transfer function from the speaker 7 to the microphone 1b has a low frequency range and a high frequency range lower than the mid range, the sound deadening effect in the microphone 1b is low. The larger the range, the greater the volume of muffling, but in the high range, the volume of muffling may be low or may not be muffled. In such a case, if it is desired to secure the high-range de-assertion level to the same extent as the low range, the equalizer 1 as shown in FIG.
3, the frequency characteristic of the transfer function from the speaker 7 to the microphone 1b is improved by increasing the level of the frequency characteristic in the higher frequency range. The muffling effect of is obtained uniformly in each band.
【００２３】このようにマイクロホン１ｂからの誤差信
号を常に監視しながらアダプティブフィルタ２の係数が
更新され続けることによって消音していくことになる
が、なんらかの要因により間違った特性になることがあ
る。するとマイクロホン１ｂで騒音付加となることがあ
る。そこで平均化手段１０では時間変動を伴うマイクロ
ホン１ｂの誤差信号を時間平均し、その出力を比較手段
１１に入力するものとする。比較手段１１では図２に示
すように平均化された誤差信号を単位時間毎に比較し
て、誤差信号が収束したかどうかを判定する、そこで誤
差信号が収束したと判定されると、停止手段１２により
ＬＭＳ演算器４によるアダプティブフィルタ２の係数更
新を停止することができる。As described above, while the error signal from the microphone 1b is constantly monitored, the coefficient of the adaptive filter 2 is continuously updated to mute the sound. However, the characteristic may be incorrect due to some factor. Then, noise may be added to the microphone 1b. Therefore, it is assumed that the averaging means 10 time-averages the error signal of the microphone 1b that is subject to time fluctuation, and the output thereof is input to the comparing means 11. The comparison means 11 compares the averaged error signals as shown in FIG. 2 every unit time to determine whether or not the error signals have converged. When it is determined that the error signals have converged, the stopping means By 12, the coefficient update of the adaptive filter 2 by the LMS calculator 4 can be stopped.
【００２４】このように本発明の第２実施例の消音装置
によれば、アダプティブフィルタ２の出力の周波数特性
を調整し消音量を確保し、誤差信号を監視することによ
り、アダプティブフィルタ２の係数更新が収束すること
を検出し、騒音付加となる前にアダプティブフィルタ２
の係数の更新を止めることができ、発散することなく安
全に騒音制御ができるものである。As described above, according to the silencer of the second embodiment of the present invention, the coefficient of the adaptive filter 2 is adjusted by adjusting the frequency characteristic of the output of the adaptive filter 2 to secure the silence volume and monitoring the error signal. The adaptive filter 2 detects that the update is converged and before noise is added.
The updating of the coefficient of can be stopped, and the noise can be safely controlled without diverging.
【００２５】つぎに本発明の第３の実施例について図４
を参照しながら説明する。図４に示すように、消音装置
は送風ダクト８内に、第１の騒音検出器であるマイクロ
ホン１ａと第２の騒音検出器であるマイクロホン１ｂ
と、スピーカ７を配し、制御装置２０内に演算器６と、
アダプティブフィルタ２と、信号処理回路であるところ
のイコライザ１３と、ＦＩＲフィルタ５と、ＦＩＲフィ
ルタ３と、アダプティブフィルタ２の係数更新を実現す
るＬＭＳ演算器４と、マイクロホン１ｂの出力を平均化
する平均化手段１０と、この平均化手段の出力を単位時
間毎に比較する比較手段１１と、アダプティブフィルタ
２の係数更新を停止する停止手段１２を配している。Next, a third embodiment of the present invention will be described with reference to FIG.
Will be described with reference to. As shown in FIG. 4, the muffler has a microphone 1a as a first noise detector and a microphone 1b as a second noise detector in the air duct 8.
And a speaker 7, and a computing unit 6 in the control device 20,
Adaptive filter 2, equalizer 13 that is a signal processing circuit, FIR filter 5, FIR filter 3, LMS calculator 4 that implements coefficient updating of adaptive filter 2, and average that averages the outputs of microphone 1b. Arrangement means 10, comparison means 11 for comparing the output of the averaging means for each unit time, and stop means 12 for stopping the coefficient update of the adaptive filter 2 are arranged.
【００２６】上記構成において、騒音は送風ダクト８内
を矢印９の方向に伝搬され、マイクロホン１ａで検出さ
れる。その検出信号と音響フィードバック補正用のＦＩ
Ｒフィルタ５の出力信号は演算器６に入力され、ＦＩＲ
フィルタ５の出力信号はマイクロホン１ａの検出信号に
対して逆位相で加算される。ここでＦＩＲフィルタ５に
は予めスピーカ７からマイクロホン１ａまでの伝達関数
が係数として近似されている。演算器６の出力はアダプ
ティブフィルタ２とＦＩＲフィルタ３に入力され、アダ
プティブフィルタ２で信号処理された信号はＦＩＲフィ
ルタ５とスピーカ７に入力され再生される。このスピー
カ７で再生された制御音とダクト内を伝搬されてくる騒
音は、マイクロホン１ｂの位置で音響的に加算され、誤
差信号としてイコライザ１３で周波数特性を調整されて
ＬＭＳ演算器４と平均化手段１０に入力される。ＬＭＳ
演算器４はこの誤差信号とＦＩＲフィルタ３からの出力
信号により、アダプティブフィルタ２の係数を演算し、
マイクロホン１ｂにおいて騒音が減衰するように係数更
新を行う。ここでＦＩＲフィルタ３には予めスピーカ７
からマイクロホン１ｂまでの伝達関数が係数として近似
されている。In the above structure, noise is propagated in the air duct 8 in the direction of the arrow 9 and detected by the microphone 1a. FI for the detection signal and acoustic feedback correction
The output signal of the R filter 5 is input to the computing unit 6 and the FIR
The output signal of the filter 5 is added in the opposite phase to the detection signal of the microphone 1a. Here, the transfer function from the speaker 7 to the microphone 1a is approximated to the FIR filter 5 in advance as a coefficient. The output of the computing unit 6 is input to the adaptive filter 2 and the FIR filter 3, and the signal processed by the adaptive filter 2 is input to the FIR filter 5 and the speaker 7 and reproduced. The control sound reproduced by the speaker 7 and the noise propagating through the duct are acoustically added at the position of the microphone 1b, and the frequency characteristics are adjusted as an error signal by the equalizer 13 and averaged with the LMS calculator 4. It is input to the means 10. LMS
The arithmetic unit 4 calculates the coefficient of the adaptive filter 2 from the error signal and the output signal from the FIR filter 3,
The coefficient is updated so that noise is attenuated in the microphone 1b. Here, the FIR filter 3 has a speaker 7 in advance.
To the microphone 1b are approximated as coefficients.
【００２７】ここでマイクロホン１ａ、１ｂで検出され
る騒音信号、誤差信号の周波数特性が低域では大きく、
高域では小さい特性を有し、スピーカ７からマイクロホ
ン１ｂまでの伝達関数の周波数特性が低域、中域に比べ
高域が落ち込んでいる特性を有している時、マイクロホ
ン１ｂにおける消音効果は低域になるほど大きな消音量
が得られるが、高域では消音量が小さいあるいは消音で
きない場合がある。このようなとき高域の消音量も低域
と同程度確保したい場合は、図３のようにイコライザ１
３を用い、その周波数特性を高域ほどレベルが大きくな
るようにすれば前記スピーカ７からマイクロホン１ｂま
での伝達関数の周波数特性は、高域の落ち込みが改善さ
れることになるので、マイクロホン１ｂでの消音効果は
各帯域で一様に得られる様になる。Here, the frequency characteristics of the noise signal and the error signal detected by the microphones 1a and 1b are large in the low range,
When the high frequency range has a small characteristic, and the frequency characteristic of the transfer function from the speaker 7 to the microphone 1b has a low frequency range and a high frequency range lower than the mid range, the sound deadening effect in the microphone 1b is low. The larger the range, the greater the volume of muffling, but in the high range, the volume of muffling may be low or may not be muffled. In such a case, if it is desired to secure the high-range de-assertion level to the same extent as the low range, the equalizer 1 as shown in FIG.
3, the frequency characteristic of the transfer function from the speaker 7 to the microphone 1b is improved by increasing the level of the frequency characteristic in the higher frequency range. The muffling effect of is obtained uniformly in each band.
【００２８】このようにマイクロホン１ｂからの誤差信
号を常に監視しながらアダプティブフィルタ２の係数が
更新され続けることによって消音していくことになる
が、なんらかの要因により間違った特性になることがあ
る。するとマイクロホン１ｂで騒音付加となることがあ
る。そこで平均化手段１０では時間変動を伴うマイクロ
ホン１ｂの誤差信号を時間平均し、その出力を比較手段
１１に入力するものとする。比較手段１１では図２に示
すように平均化された誤差信号を単位時間毎に比較し
て、誤差信号が収束したかどうかを判定する、そこで誤
差信号が収束したと判定されると、停止手段１２により
ＬＭＳ演算器４によるアダプティブフィルタ２の係数更
新を停止することができる。As described above, while the error signal from the microphone 1b is constantly monitored, the coefficient of the adaptive filter 2 is continuously updated to mute the sound. However, the characteristic may be incorrect due to some factor. Then, noise may be added to the microphone 1b. Therefore, it is assumed that the averaging means 10 time-averages the error signal of the microphone 1b that is subject to time fluctuation, and the output thereof is input to the comparing means 11. The comparison means 11 compares the averaged error signals as shown in FIG. 2 every unit time to determine whether or not the error signals have converged. When it is determined that the error signals have converged, the stopping means By 12, the coefficient update of the adaptive filter 2 by the LMS calculator 4 can be stopped.
【００２９】このように本発明の第３実施例の消音装置
によれば、アダプティブフィルタ２の出力の周波数特性
を調整し消音量を確保し、誤差信号を監視することによ
り、アダプティブフィルタ２の係数更新が収束すること
を検出し、騒音付加となる前にアダプティブフィルタ２
の係数の更新を止めることができ、発散することなく安
全に騒音制御ができるものである。As described above, according to the silencer of the third embodiment of the present invention, the coefficient of the adaptive filter 2 is adjusted by adjusting the frequency characteristic of the output of the adaptive filter 2 to secure the silence volume and monitoring the error signal. The adaptive filter 2 detects that the update is converged and before noise is added.
The updating of the coefficient of can be stopped, and the noise can be safely controlled without diverging.
【００３０】[0030]
【発明の効果】以上の実施例から明らかなように本発明
によれば、誤差信号の時間毎の平均値を比較することに
より、アダプティブフィルタの係数更新が収束した時点
で係数更新を停止する停止手段を用いたことにより、騒
音付加とならず最も消音効果のあるアダプティブフィル
タの係数で安全に騒音制御ができる消音装置が提供でき
る。As is apparent from the above embodiments, according to the present invention, by comparing the average values of error signals for each time, the stop of the coefficient update is stopped when the coefficient update of the adaptive filter converges. By using the means, it is possible to provide a noise reduction device that can perform noise control safely with the coefficient of the adaptive filter having the most noise reduction effect without adding noise.
【００３１】また、アダプティブフィルタの出力や誤差
信号の周波数特性を調整する信号処理回路のより、周波
数特性が平坦でない騒音に対しても各周波数帯で一様に
消音可能となり、前記停止手段を用いることにより騒音
付加とならず最も消音効果のあるアダプティブフィルタ
の係数で安全に騒音制御ができる消音装置が提供でき
る。Further, since the signal processing circuit for adjusting the frequency characteristic of the output of the adaptive filter and the error signal is used, the noise can be silenced uniformly in each frequency band even for the noise whose frequency characteristic is not flat, and the stopping means is used. As a result, it is possible to provide a noise reduction device that can safely perform noise control with the coefficient of the adaptive filter having the most noise reduction effect without adding noise.
【図１】本発明の第１実施例の消音装置のブロック図FIG. 1 is a block diagram of a silencer according to a first embodiment of the present invention.
【図２】同第１実施例の消音装置の誤差信号の平均値の
時間特性図FIG. 2 is a time characteristic diagram of an average value of error signals of the silencer of the first embodiment.
【図３】同第２実施例の消音装置のブロック図FIG. 3 is a block diagram of a silencer according to the second embodiment.
【図４】同第３実施例の消音装置のブロック図FIG. 4 is a block diagram of a silencer according to the third embodiment.
【図５】従来の消音装置のブロック図FIG. 5 is a block diagram of a conventional silencer.
１ａ、１ｂ マイクロホン ２ アダプティブフィルタ ３ ＦＩＲフィルタ ４ ＬＭＳ演算器 ５ ＦＩＲフィルタ ６ 演算器 ７ スピーカ ８ 送風ダクト ９ 騒音の伝搬する方向 １０ 平均化手段 １１ 比較手段 １２ 停止手段 ２０ 制御装置 1a, 1b Microphone 2 Adaptive filter 3 FIR filter 4 LMS calculator 5 FIR filter 6 calculator 7 Speaker 8 Air duct 9 Noise propagation direction 10 Averager 11 Comparing means 12 Stopping means 20 Control device
Claims (3)
音源として伝搬する騒音を検出する第１の騒音検出機
と、前記第１の騒音検出器の出力を信号処理する演算器
と、前記演算器の出力を適応制御するアダプティブフィ
ルタと、前記演算器の出力を信号処理する第１のディジ
タルフィルタと、前記第１の騒音検出器の設置位置に比
べて騒音源から遠い位置に設置され、前記アダプティブ
フィルタの出力を再生するスピーカと、前記アダプティ
ブフィルタの出力を信号処理してその結果を前記演算器
に入力する第２のディジタルフィルタと、前記スピーカ
の近傍の騒音源から遠い位置に設置された第２の騒音検
出器と、前記第２の騒音検出器の出力と前記ディジタル
フィルタの出力から前記アダプティブフィルタの時間領
域における係数を演算して更新する係数演算器と、前記
第２の騒音検出器の出力を演算して時間領域における平
均値を算出する平均化手段と、前記平均化手段の出力を
単位時間毎に比較し前記出力が収束したことを検出する
比較手段と、前記比較手段の出力から前記係数演算器に
よる係数更新を停止する停止手段を配してなる消音装
置。1. A first noise detector for detecting noise propagating with an air outlet or an air inlet of an air conditioner as a noise source, a calculator for processing the output of the first noise detector, and An adaptive filter that adaptively controls the output of the arithmetic unit, a first digital filter that performs signal processing of the output of the arithmetic unit, and a farther from a noise source than the installation position of the first noise detector, A speaker for reproducing the output of the adaptive filter, a second digital filter for signal-processing the output of the adaptive filter and inputting the result to the arithmetic unit, and a speaker installed near the speaker and far from a noise source. A second noise detector, and a coefficient in the time domain of the adaptive filter is calculated from the output of the second noise detector and the output of the digital filter. And a coefficient calculator for updating, an averaging means for computing the output of the second noise detector to calculate an average value in the time domain, and an output for comparing the output of the averaging means for each unit time. A silencer comprising a comparison means for detecting that the coefficient has converged and a stop means for stopping the coefficient update by the coefficient calculator from the output of the comparison means.
性を調整し、その結果を前記スピーカに出力する信号処
理回路を配してなる請求項１記載の消音装置。2. The silencer according to claim 1, further comprising a signal processing circuit for adjusting the frequency characteristic of the output of the adaptive filter and outputting the result to the speaker.
調整し、その結果を係数演算器に出力する信号処理回路
を配してなる請求項１記載の消音装置。3. The silencer according to claim 1, further comprising a signal processing circuit for adjusting the frequency characteristic of the output of the second noise detector and outputting the result to the coefficient calculator.
Priority Applications (1)
| Application Number | Priority Date | Filing Date | Title |
|---|---|---|---|
| JP6121171A JPH07325588A (en) | 1994-06-02 | 1994-06-02 | Muffler |
Applications Claiming Priority (1)
| Application Number | Priority Date | Filing Date | Title |
|---|---|---|---|
| JP6121171A JPH07325588A (en) | 1994-06-02 | 1994-06-02 | Muffler |
Publications (1)
| Publication Number | Publication Date |
|---|---|
| JPH07325588A true JPH07325588A (en) | 1995-12-12 |
Family
ID=14804610
Family Applications (1)
| Application Number | Title | Priority Date | Filing Date |
|---|---|---|---|
| JP6121171A Pending JPH07325588A (en) | 1994-06-02 | 1994-06-02 | Muffler |
Country Status (1)
| Country | Link |
|---|---|
| JP (1) | JPH07325588A (en) |
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-
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