JP3217614B2 - Acoustic echo canceller - Google Patents

Acoustic echo canceller

Info

Publication number
JP3217614B2
JP3217614B2 JP23486194A JP23486194A JP3217614B2 JP 3217614 B2 JP3217614 B2 JP 3217614B2 JP 23486194 A JP23486194 A JP 23486194A JP 23486194 A JP23486194 A JP 23486194A JP 3217614 B2 JP3217614 B2 JP 3217614B2
Authority
JP
Japan
Prior art keywords
coefficient
signal input
reception signal
input terminal
acoustic
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.)
Expired - Fee Related
Application number
JP23486194A
Other languages
Japanese (ja)
Other versions
JPH0897752A (en
Inventor
吉雅 草野
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.)
Kyocera Corp
Original Assignee
Kyocera Corp
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 Kyocera Corp filed Critical Kyocera Corp
Priority to JP23486194A priority Critical patent/JP3217614B2/en
Publication of JPH0897752A publication Critical patent/JPH0897752A/en
Application granted granted Critical
Publication of JP3217614B2 publication Critical patent/JP3217614B2/en
Anticipated expiration legal-status Critical
Expired - Fee Related legal-status Critical Current

Links

Landscapes

  • Cable Transmission Systems, Equalization Of Radio And Reduction Of Echo (AREA)

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、通信回線、室内音場制
御装置そして高品質な音声通信会議装置に使用され、受
話径路の信号が音響反響経路を介して送話経路に現れる
音響反響成分を除去する音響反響除去装置に関する。
BACKGROUND OF THE INVENTION 1. Field of the Invention The present invention is used for a communication line, an indoor sound field control device, and a high-quality audio communication conference device, and a signal of a receiving path appears on a transmission path via an acoustic reflection path. The present invention relates to an acoustic reverberation removing device for removing noise.

【0002】[0002]

【従来の技術】一般に音響反響除去装置は通信衛生およ
び海底ケーブルを利用した長距離電話回線において、2
線4線変換器のインピーダンス不整合により生ずる反射
を除去するものと、テレビ会議システムなどの拡声電話
において、話者音声の音響結合による反響を除去するも
のとに大別でき、修正量演算回路、擬似音響反響を発生
する可変係数フィルタおよび減算回路から構成されてい
る。以下に音響反響除去装置の基本動作を述べる。
2. Description of the Related Art In general, acoustic echo cancellers are used for long-distance telephone lines using communication sanitation and submarine cables.
It can be broadly divided into one that removes reflection caused by impedance mismatch of the line-to-wire converter and one that removes reverberation due to acoustic coupling of speaker's voice in a loudspeaker such as a video conference system. It is composed of a variable coefficient filter for generating a pseudo acoustic reverberation and a subtraction circuit. The basic operation of the acoustic reverberation removing device will be described below.

【0003】図5は音響反響除去装置の基本構成を示す
図である。受話信号入力端子1は受話信号出力端子2に
接続され、その受話信号入力端子1の受話信号は可変係
数フィルタ3に分岐供給され、擬似音響反響を生成させ
る。送話信号入力端子4からの送話信号と可変係数フィ
ルタ3の出力である擬似音響反響は減算回路5へ入力さ
れ、送話信号中の音響反響成分が除去され、その減算回
路5の出力は送話信号出力端子6へ出力される。送話信
号出力端子6の出力と受話信号入力端子1の信号が修正
量演算回路7に入力され、係数修正量演算回路7の出力
により可変係数フィルタ3のフィルタ係数が修正され
る。可変係数フィルタ3内で受話信号は受話信号入力レ
ジスタ8に入力され、その受話信号入力レジスタ8の受
話信号と擬似インパルス応答レジスタ9の擬似インパル
ス応答との積和が積和回路10でとられ、積和回路10
の出力が擬似音響反響として出力される。受話信号出力
端子2および送話信号入力端子4は長距離電話回線の場
合、2線4線変換器に、拡声電話システムの場合、スピ
ーカとマイクロホンへと接続されている。
FIG. 5 is a diagram showing a basic configuration of an acoustic reverberation removing apparatus. The reception signal input terminal 1 is connected to the reception signal output terminal 2, and the reception signal of the reception signal input terminal 1 is branched and supplied to the variable coefficient filter 3 to generate a pseudo acoustic echo. The transmission signal from the transmission signal input terminal 4 and the pseudo acoustic reverberation output from the variable coefficient filter 3 are input to a subtraction circuit 5, where the acoustic reverberation component in the transmission signal is removed, and the output of the subtraction circuit 5 is It is output to the transmission signal output terminal 6. The output of the transmission signal output terminal 6 and the signal of the reception signal input terminal 1 are input to the correction amount calculation circuit 7, and the filter coefficient of the variable coefficient filter 3 is corrected by the output of the coefficient correction amount calculation circuit 7. In the variable coefficient filter 3, the reception signal is input to the reception signal input register 8, and the product sum of the reception signal of the reception signal input register 8 and the pseudo impulse response of the pseudo impulse response register 9 is obtained by the product sum circuit 10. Product-sum circuit 10
Is output as a pseudo acoustic echo. The reception signal output terminal 2 and the transmission signal input terminal 4 are connected to a two-wire / four-wire converter for a long-distance telephone line, and to a speaker and a microphone for a loudspeaker system.

【0004】音響反響経路の信号伝搬特性を線形で、且
つFIR形ディジタルフィルタで表されると仮定し、そ
のインパルス応答h(t)と入力受話信号x(t)とを
用いれば、サンプル時間間隔をTとし、時刻kTにおけ
る音響反響yk は、 yk = h’xk (1) で表される。但し、 h=[h1 ,h2 ,・・・,hn ]’ (2) x=[xk-1 ,・・・,xk-n ]’ ’:べクトルの転置である。
[0004] Assuming that the signal propagation characteristics of the acoustic reverberation path are linear and represented by an FIR type digital filter, and using the impulse response h (t) and the input received signal x (t), a sample time interval is obtained. was T, the acoustic echo yk is at time kT, represented by y k = h'x k (1) . Here, h = [h 1 , h 2 ,..., H n ] ′ ′ (2) x = [x k−1 ,..., X kn ] ′ ′: Vector transposition.

【0005】一方、 時刻kTにおけるhの推定値をh
k とすれば、yk の推定値yskは、 ysk = hsk ’xk (3) で与えられる。 音響反響除去装置では、受話信号入力
端子1に音声信号があり、送話信号入力端子4に音声信
号がなく音響反響のみが存在している時、適応動作状態
として反響除去動作を行う。この適応動作アルゴリズム
には、一般に学習同定法(野田淳彦、南雲仁一:“シス
テムの学習同定法”計測と制御、7、9、pp.597-605(1
968))が採用される。学習同定法によるhsk の逐次修
正は、 hsk+1 = hsk +α(xk k )/xk ’xk (4)に よって行われる。但し、 ek =yk −ysk , 0<α≦1 (5) でありek を残留音響反響と呼ぶ。この様な演算動作が
係数修正量演算回路7において処理実行されている。擬
似インパルス応答レジスタ9の内容には可変係数系列h
sk が格納されている。αは推定の敏感さを決定する為
の係数更新利得で1.0に近いほど大きな修正量を与え
る事ができ、高速な音響反響除去が可能となるが、実際
に用いる場合には近端雑音や回線状態によって変えて設
定する必要がある。この係数更新利得αの決定は、現在
のところ経験則に依っているのが実態である。又、この
係数更新利得αを残留音響反響の大きさにより可変制御
するものや室内特性に合わせて設定するものがある(例
えば、牧野昭二、小泉宣夫:“エコーキャンセラの室内
音場における適応特性の改善について”、信学論
(A)、J71-A,12,pp.2212-2214(1988-12))。
On the other hand, the estimated value of h at time kT is expressed as h
if s k, estimated value ys k of y k is given by ys k = hs k 'x k (3). In the acoustic reverberation removing device, when there is an audio signal at the receiving signal input terminal 1 and no acoustic signal exists at the transmitting signal input terminal 4 and only acoustic reverberation exists, the acoustic reverberation operation is performed as an adaptive operation state. This adaptive operation algorithm generally includes a learning identification method (Atsuhiko Noda, Jinichi Nagumo: “System Learning Identification Method”, Measurement and Control, 7, 9, pp. 597-605 (1
968)). Successive correction of hsk by the learning identification method is performed by the hs k + 1 = hs k + α (x k e k) / x k 'x k (4). However, the e k = y k -ys k, 0 < a α ≦ 1 (5) ek called the residual acoustic echo. Such a calculation operation is performed in the coefficient correction amount calculation circuit 7. The contents of the pseudo impulse response register 9 include a variable coefficient series h.
sk is stored. α is a coefficient updating gain for determining the sensitivity of estimation, and the closer the correction gain is to 1.0, the greater the amount of correction can be given, and high-speed acoustic reverberation can be removed. It is necessary to change and set according to the line condition. The actual situation is that the determination of the coefficient update gain α currently depends on empirical rules. The coefficient update gain α may be variably controlled according to the magnitude of the residual acoustic reverberation, or may be set in accordance with the room characteristics (for example, Shoji Makino, Nobuo Koizumi: “The adaptation characteristics of the echo canceller in the room sound field. Improvement, "IEICE (A), J71-A, 12, pp. 221-2214 (1988-12)).

【0006】[0006]

【発明が解決しようとする課題】最小二乗法(LMS)
を基本とする学習同定法によるパラメータ推定では、係
数更新利得αの量にその推定性能は大きく依存してい
る。式(5)よりαの取る範囲は0から1の間に有れば
それなりの性能は得られるのだが、その値の差により収
束速度と飽和反響除去量が異なってくる。その状況を示
したのが図4で係数更新利得αの値は図中aが1.0と
図中bが0.5の時のものである。α値が大きいほど高
速になるが飽和反響除去量は逆に低下する事が判る。高
速化と動作安定化はトレードオフの関係にあり、従っ
て、高速性と動作安定性を両立させるのは困難であると
いう問題点があった。
SUMMARY OF THE INVENTION Least squares method (LMS)
In the parameter estimation by the learning identification method based on, the estimation performance greatly depends on the amount of the coefficient update gain α. According to equation (5), a certain performance can be obtained if the range of α is between 0 and 1, but the convergence speed and the amount of saturation reverberation differ depending on the difference between the values. FIG. 4 shows the situation when the value of the coefficient update gain α is 1.0 in the figure and 0.5 in the figure. It can be seen that the higher the value of α, the faster the speed, but the amount of saturation reverberation decreases. There is a trade-off between speeding up and operation stabilization, and there is a problem that it is difficult to achieve both high speed and operation stability.

【0007】本発明は上述の点に鑑みてなされたもの
で、上記問題点を除去し、高速性と動作安定性に優れ、
高い適応性能を有し、常時大きな音響反響消去量を維持
しながら音響制御を行う音響反響除去装置を提供する事
を目的とする。
[0007] The present invention has been made in view of the above points, and eliminates the above-mentioned problems, and is excellent in high-speed operation and operation stability.
An object of the present invention is to provide an acoustic reverberation removing apparatus which has high adaptive performance and performs acoustic control while always maintaining a large acoustic echo canceling amount.

【0008】[0008]

【課題を解決するための手段】本発明はこれらの課題を
解決するためのものであり、受話信号入力端子と、受話
信号出力端子と、送話信号入力端子と、送話信号出力端
子と、前記受話信号入力端子から入力とする擬似インパ
ルス応答レジスタを有する可変係数デジタルフィルタ
と、前記受話信号出力端子から音響反響経路を介して
送話信号入力端子に入力される受話信号の音響反響成
分から前記可変係数デジタルフィルタで生起された擬似
音響反響を減算して求められる残差信号を最小とするよ
うに係数修正量演算回路によって係数系列が逐次更新さ
れる音響反響除去装置において、前記擬似インパルス応
答レジスタに格納された各可変係数の累積加算平均電力
の変動率の値を回路に内挿された数種類の各閾値と比較
し、一定値以下になったかどうかによって更新演算処理
対象ブロックであるかを選択することができる係数更新
利得選択回路を有する音響反響除去装置を提供する。
SUMMARY OF THE INVENTION The present invention has been made to solve these problems, and includes a reception signal input terminal, a reception signal output terminal, a transmission signal input terminal, a transmission signal output terminal, and a variable coefficient digital filter having a pseudo impulse response register which receives from the reception signal input terminal, via the front acoustic echo path from said reception signal output terminal
A coefficient correction amount calculating circuit is used to minimize the residual signal obtained by subtracting the pseudo acoustic reverberation generated by the variable coefficient digital filter from the acoustic reverberant component of the received signal input to the transmission signal input terminal. In the acoustic reverberation apparatus in which the coefficient sequence is sequentially updated, the value of the variation rate of the cumulative average power of each variable coefficient stored in the pseudo impulse response register is compared with several types of threshold values inserted in the circuit, Provided is an acoustic reverberation removal apparatus having a coefficient update gain selection circuit capable of selecting whether a block is an update operation processing target depending on whether the block becomes a predetermined value or less.

【0009】[0009]

【作用】本発明では、上記手段により推定動作の高速性
と高安定性が確保されるので、通信回線上に反響成分が
混入する事が極めて少なくなり、通信音声音質の劣化を
防ぎ、通話そのものを出来なくしてしまうハウリング発
生の危険性を低く抑える事が出来、高品質な音響制御が
可能となる。
According to the present invention, since the high speed and high stability of the estimating operation are ensured by the above-mentioned means, reverberation components are hardly mixed into the communication line, the deterioration of the sound quality of the communication voice is prevented, and the communication itself is prevented. This can reduce the risk of howling occurrence, which makes it impossible to perform sound control, and enables high-quality sound control.

【0010】[0010]

【実施例】以下本発明の実施例を図面にもとづいて詳細
に説明する。図1は本発明の音響反響除去装置の構成を
示すブロック図である。図1に示されるように、本発明
は従来の受話信号入力端子1、3受話信号出力端子2、
可変係数ディジタルフィルタ3、送話信号入力端子4、
減算回路5、送話信号出力端子6、係数修正量演算回路
7、受話信号入力レジスタ8、擬似インパルス応答レジ
スタ9、そして、積和演算回路10から構成された適応
アルゴリズムとして学習同定法を採用した音響反響除去
装置と同一構成の装置に、累積加算平均電力演算回路1
1、電力変動率算出回路12、そして、係数更新利得選
択回路13を追加した構成となっている。該受話信号入
力端子1と、該受話信号出力端子2と、該送話信号入力
端子4と、該送話信号出力端子6と、該受話信号入力端
子1の受話信号を入力とする該擬似インパルス応答レジ
スタ9を持つ該可変係数ディジタルフィルタ3と、該受
話信号出力端子2から音響反響経路を介して該送話信号
入力端子4に入力される受話信号の音響反響成分から該
可変係数ディジタルフィルタ3で生起された擬似音響反
響を減算して求められる残差信号を最小とする様に該係
数修正量演算回路7によって係数系列が逐次更新される
音響反響除去装置において、該擬似インパルス応答レジ
スタ9に格納された各可変係数単独のの累積加算平均電
力を算出する該累積加算平均電力演算回路11と、各可
変係数単独の累積加算平均電力の変動率を算出する該電
力変動率算出回路12と、累積加算平均電力の変動率に
対応した該係数更新利得選択回路13に記憶された0.
0から1.0の範囲の係数更新利得αを選択して該係数
修正量演算回路7にその選択値を送出し、この値を基に
式(4)によって修正量を算出する事を特徴とする音響
反響除去装置。該係数更新利得選択回路13では、係数
電力の変動率の値を回路に内挿された数種類の各閾値と
比較し、適合する閾値範囲に対応させた係数更新利得を
選択する動作を行っている。図2にこの概念図を示す。
この時、係数電力の変動率が一定以下になったと検出し
た場合には、その係数の該係数修正量演算回路7での修
正量演算と更新演算を行わない設定にしておけば、演算
量を削減さす事が可能となり、ハードウェアの負担が軽
減できる。変動率の算出法は以下の通りである。
DETAILED DESCRIPTION OF THE PREFERRED EMBODIMENTS Embodiments of the present invention will be described below in detail with reference to the drawings. FIG. 1 is a block diagram showing a configuration of an acoustic reverberation removing apparatus according to the present invention. As shown in FIG. 1, the present invention provides a conventional reception signal input terminal 1, a conventional reception signal output terminal 2,
Variable coefficient digital filter 3, transmission signal input terminal 4,
A learning identification method is adopted as an adaptive algorithm comprising a subtraction circuit 5, a transmission signal output terminal 6, a coefficient correction amount operation circuit 7, a reception signal input register 8, a pseudo impulse response register 9, and a product-sum operation circuit 10. An accumulative average power calculation circuit 1
1, a power fluctuation rate calculation circuit 12, and a coefficient update gain selection circuit 13 are added. The pseudo-impulse which receives the reception signal input terminal 1, the reception signal output terminal 2, the transmission signal input terminal 4, the transmission signal output terminal 6, and the reception signal of the reception signal input terminal 1. The variable coefficient digital filter 3 having a response register 9 and the variable coefficient digital filter 3 based on the acoustic reverberation component of the received signal input from the received signal output terminal 2 to the transmission signal input terminal 4 via the acoustic reverberation path. In the acoustic reverberation removing apparatus in which the coefficient sequence is sequentially updated by the coefficient correction amount calculating circuit 7 so as to minimize the residual signal obtained by subtracting the pseudo acoustic reverberation generated in the above, the pseudo impulse response register 9 The accumulated average power calculation circuit 11 for calculating the accumulated average power of each of the stored variable coefficients alone, and the power conversion circuit 11 calculating the variation rate of the accumulated average power of each of the variable coefficients alone. A rate calculation circuit 12, stored in the coefficient update gain selection circuit 13 corresponding to the variation rate of the cumulative addition average power 0.
A coefficient update gain α in the range of 0 to 1.0 is selected, the selected value is sent to the coefficient correction amount calculation circuit 7, and the correction amount is calculated based on this value by equation (4). Acoustic echo canceller. The coefficient update gain selection circuit 13 performs an operation of comparing the value of the variation rate of the coefficient power with each of several threshold values inserted in the circuit and selecting a coefficient update gain corresponding to a suitable threshold range. . FIG. 2 shows this conceptual diagram.
At this time, if it is detected that the variation rate of the coefficient power has become equal to or less than a certain value, if the correction amount calculation and the update calculation of the coefficient in the coefficient correction amount calculation circuit 7 are not performed, the calculation amount is reduced. It is possible to reduce the load and reduce the load on the hardware. The calculation method of the fluctuation rate is as follows.

【0011】 dhk+1 =|hsk+1 2−hsk 2 |/hsk 2 (6) 式(6)に示した変動率は一例であり、決定的な算出法
ではない。例えば分子が過去値ではなくて現在値でもよ
い。又、dhk+1 とdhk との間での差分値を用いても
本発明は有効に機能する。
[0011] dh k + 1 = | hs k + 1 2 -hs k 2 | / hs k 2 (6) fluctuation rate shown in equation (6) is an example, not a definitive calculation method. For example, the numerator may be a present value instead of a past value. Also, the present invention functions effectively even if a difference value between dhk + 1 and dhk is used.

【0012】図3に白色雑音を参照入力とした場合の本
発明による適応処理動作の結果aを示す。比較対象とし
て係数更新利得を0.5に固定したモデルの結果bを同
図に載せている。縦軸は音響反響消去量、横軸は時間で
ある。本発明によるモデルの係数更新利得の最大設定値
は1.0、最小設定値は0.05とした。初期の消去過
渡領域における応答速度は音響反響消去量が30[d
B]の時で比較すると約2倍本発明によるモデルの方が
優れている事が判る。そして、係数更新利得が徐々に小
さな値に設定されているので外乱の影響を受けずに済
む。
FIG. 3 shows a result a of the adaptive processing operation according to the present invention when white noise is used as a reference input. As a comparison target, the result b of the model in which the coefficient update gain is fixed to 0.5 is shown in FIG. The vertical axis is the amount of acoustic echo cancellation, and the horizontal axis is time. The maximum set value of the coefficient update gain of the model according to the present invention was 1.0, and the minimum set value was 0.05. The response speed in the initial erasure transient region is 30 [d
B], it can be seen that the model according to the present invention is approximately twice as good as the model according to the present invention. Since the coefficient update gain is gradually set to a small value, the coefficient update gain is not affected by disturbance.

【0013】[0013]

【発明の効果】以上、詳細に説明したように本発明によ
れば、下記のような優れた効果が期待される。
As described above, according to the present invention, the following excellent effects are expected.

【0014】(1)本発明を用いる事で、高速化と高安
定化を同時に実現できるので、高品質な音声通信の維持
を図れ、ハウリング発生の危険性を低く抑える事ができ
る。 (2)適応動作過程において、更新演算を行わずに済む
可変係数が発生する。この事により演算量が削減され、
ハードウェアの負担が低減出来る。
(1) By using the present invention, high speed and high stability can be realized at the same time, so that high quality voice communication can be maintained, and the danger of howling can be suppressed. (2) In the adaptive operation process, a variable coefficient that does not need to be updated is generated. This reduces the amount of computation,
Hardware burden can be reduced.

【0015】(3)必要以上に可変係数の更新を行わな
いのでディジタルシグナルプロセッサ等で構成する場
合、演算誤差や誤動作を生じにくい。
(3) Since the variable coefficients are not updated more than necessary, when a digital signal processor or the like is used, calculation errors and malfunctions are less likely to occur.

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

【図1】 本発明による音響反響除去装置の一構成例を
示すブロック図である。
FIG. 1 is a block diagram showing a configuration example of an acoustic reverberation removing apparatus according to the present invention.

【図2】 本説明に用いた状態判定制御部の概念を示す
ブロック図である。
FIG. 2 is a block diagram illustrating a concept of a state determination control unit used in the description.

【図3】 本発明に用いた本発明による白色雑音を参照
入力とした場合の音響反響消去特性の一例を示した図で
ある。
FIG. 3 is a diagram illustrating an example of an acoustic reverberation elimination characteristic when white noise according to the present invention used in the present invention is used as a reference input;

【図4】 本説明に用いた本発明による白色雑音を参照
入力とした場合の従来方式による音響反響消去特性の一
例を示した図である。
FIG. 4 is a diagram showing an example of acoustic echo cancellation characteristics according to a conventional method when the white noise according to the present invention used in the present description is used as a reference input.

【図5】 従来の一般的な学習同定法を用いた音響反響
除去装置の基本構成の一例を示したブロック図である。
FIG. 5 is a block diagram showing an example of a basic configuration of a conventional acoustic reverberation removing apparatus using a general learning identification method.

【符号の説明】[Explanation of symbols]

1 受話信号入力端子 2 受話信号出力端子 3 可変係数フィルタ 4 送話信号入力端子 5 減算回路 6 送話信号出力端子 7 修正量演算回路 8 受話信号入力レジスタ 9 擬似インパルス応答レジスタ 10 積和演算回路 11 累積加算平均電力演算回路 12 電力変動率算出回路 13 係数更新利得選択回路 REFERENCE SIGNS LIST 1 reception signal input terminal 2 reception signal output terminal 3 variable coefficient filter 4 transmission signal input terminal 5 subtraction circuit 6 transmission signal output terminal 7 correction amount operation circuit 8 reception signal input register 9 pseudo impulse response register 10 product sum operation circuit 11 Cumulative average power calculation circuit 12 Power fluctuation rate calculation circuit 13 Coefficient update gain selection circuit

Claims (1)

(57)【特許請求の範囲】(57) [Claims] 【請求項1】 受話信号入力端子と、受話信号出力端子
と、送話信号入力端子と、送話信号出力端子と、前記受
話信号入力端子から入力とする擬似インパルス応答レジ
スタを有する可変係数デジタルフィルタと、前記受話信
号出力端子から音響反響経路を介して前記送話信号入力
端子に入力される受話信号の音響反響成分から前記可変
係数デジタルフィルタで生起された擬似音響反響を減算
して求められる残差信号を最小とするように係数修正量
演算回路によって係数系列が逐次更新される音響反響除
去装置において、前記擬似インパルス応答レジスタに格
納された各可変係数の累積加算平均電力の変動率の値を
回路に内挿された数種類の各閾値と比較し、一定値以下
になったかどうかによって更新演算処理対象ブロックで
あるかを選択することができる係数更新利得選択回路を
有することを特徴とする音響反響除去装置
1. A variable coefficient digital filter having a reception signal input terminal, a reception signal output terminal, a transmission signal input terminal, a transmission signal output terminal, and a pseudo impulse response register input from the reception signal input terminal. When the residual obtained by subtracting the pseudo acoustic echo that is arising from the acoustic echo components of the received signal input from the reception signal output terminal to the transmission signal input terminal via the acoustic echo path in said variable coefficient digital filter In an acoustic reverberation removing apparatus in which a coefficient sequence is sequentially updated by a coefficient correction amount calculating circuit so as to minimize a difference signal, the value of the variation rate of the cumulative addition average power of each variable coefficient stored in the pseudo impulse response register is calculated.
Comparison with each of several threshold values interpolated in the circuit, below a certain value
Depending on whether or not
A coefficient update gain selection circuit that can select
Acoustic echo canceller characterized by having
JP23486194A 1994-09-29 1994-09-29 Acoustic echo canceller Expired - Fee Related JP3217614B2 (en)

Priority Applications (1)

Application Number Priority Date Filing Date Title
JP23486194A JP3217614B2 (en) 1994-09-29 1994-09-29 Acoustic echo canceller

Applications Claiming Priority (1)

Application Number Priority Date Filing Date Title
JP23486194A JP3217614B2 (en) 1994-09-29 1994-09-29 Acoustic echo canceller

Publications (2)

Publication Number Publication Date
JPH0897752A JPH0897752A (en) 1996-04-12
JP3217614B2 true JP3217614B2 (en) 2001-10-09

Family

ID=16977501

Family Applications (1)

Application Number Title Priority Date Filing Date
JP23486194A Expired - Fee Related JP3217614B2 (en) 1994-09-29 1994-09-29 Acoustic echo canceller

Country Status (1)

Country Link
JP (1) JP3217614B2 (en)

Families Citing this family (1)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
KR100436479B1 (en) * 2000-03-08 2004-06-24 엘지전자 주식회사 Astriction-speed Variable Control Method Of Adaptive Echo Cancelier In Switching System

Also Published As

Publication number Publication date
JPH0897752A (en) 1996-04-12

Similar Documents

Publication Publication Date Title
JPH08288890A (en) Method and device for adaptive type filtering
JPH08265223A (en) Adaptive filter and echo canceller
JPH06113027A (en) Echo eliminating device
JP3211884B2 (en) Acoustic echo canceller
JP3217614B2 (en) Acoustic echo canceller
US5987143A (en) Method and apparatus for erasing acoustic echo
JP3217618B2 (en) Acoustic echo canceller
JP3121998B2 (en) Acoustic echo canceller
JP3121997B2 (en) Acoustic echo canceller
JP3217619B2 (en) Acoustic echo canceller
JP3152815B2 (en) Acoustic echo canceller
JPH0946276A (en) Public-address information communication system
JP3152822B2 (en) Acoustic echo canceller
JP3152825B2 (en) Acoustic echo canceller
JP3452341B2 (en) Echo canceller
JP3121983B2 (en) Acoustic echo canceller
JPH1013310A (en) Echo canceller
JP2000252884A (en) Adaptive filter learning system
JP3121988B2 (en) Acoustic echo canceller
JP3121969B2 (en) Acoustic echo canceller
JP2551869B2 (en) Echo canceller
JPH0964792A (en) Loud speaker information communication system
JPH0993166A (en) Loudspeaking information communication system
JPH07212278A (en) Echo erasing device
JP3293706B2 (en) Echo canceler

Legal Events

Date Code Title Description
FPAY Renewal fee payment (event date is renewal date of database)

Free format text: PAYMENT UNTIL: 20080803

Year of fee payment: 7

LAPS Cancellation because of no payment of annual fees