JPH08125589A - Acoustic echo cancelling device - Google Patents

Acoustic echo cancelling device

Info

Publication number
JPH08125589A
JPH08125589A JP26363894A JP26363894A JPH08125589A JP H08125589 A JPH08125589 A JP H08125589A JP 26363894 A JP26363894 A JP 26363894A JP 26363894 A JP26363894 A JP 26363894A JP H08125589 A JPH08125589 A JP H08125589A
Authority
JP
Japan
Prior art keywords
coefficient
acoustic echo
block
reception signal
fluctuation rate
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.)
Granted
Application number
JP26363894A
Other languages
Japanese (ja)
Other versions
JP3217618B2 (en
Inventor
Yoshimasa Kusano
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 JP26363894A priority Critical patent/JP3217618B2/en
Publication of JPH08125589A publication Critical patent/JPH08125589A/en
Application granted granted Critical
Publication of JP3217618B2 publication Critical patent/JP3217618B2/en
Anticipated expiration legal-status Critical
Expired - Fee Related legal-status Critical Current

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Abstract

PURPOSE: To permit high-speed property to be compatible with high stability and to enable acoustic controlling by dividing a pseudo impulse responce register into N-number of blocks, calculating the representative fluctuation rate of coefficient accumulative addition average power at every block and changing coefficient updating gain by means of the representative fluctuation rate. CONSTITUTION: An accumulative addition average power arithmetic circuit 11 calculates the single accumulative addition average power of respective variable coefficients which are stored in the pseudo impulse responce register 9 divided into N-number blocks. A block average accumulative power arithmetic circuit 12 calculates the individual average addition average power of the respective blocks. A block power fluctuation rate calculating circuit 13 calculates the fluctuation rate of the independent average accumulative addition average power of the respective blocks. A block coefficient updating gain selecting circuit 14 selects coefficient updating gain αwithin the range of zero to one corresponding to the fluctuation rate of the average accumulative addition average power of the respective blocks every time and transmits it to a coefficient correcting quantity arithmetic circuit 7. The circuit 7 calculates correcting quantity by an expression I through the use of the value. In the expression I, hSK is the estimated value of impulse responce (h) at a time kT, ek is residual acoustic echo and Xk is an input call receiving signal.

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 in a communication line, a room sound field control device, and a high-quality voice communication conference device, and an acoustic echo component in which a signal on a receiving path appears in a transmitting path via an acoustic echo path. The present invention relates to an acoustic echo canceller that removes noise.

【0002】[0002]

【従来の技術】一般に、音響反響除去装置は通信衛星お
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 Generally, an acoustic echo canceller is used in a long-distance telephone line using a communication satellite and a submarine cable.
A correction amount calculation circuit can be roughly divided into a device that removes reflection caused by impedance mismatch of a 2-wire to 4-wire converter and a device that removes reverberation due to acoustic coupling of speaker's voice in a loudspeaker telephone such as a video conference system. , A variable coefficient filter for generating pseudo acoustic echo and a subtraction circuit. The basic operation of the acoustic echo canceller 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 the basic structure of an acoustic echo canceller. 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 pseudo acoustic echo. The transmission signal from the transmission signal input terminal 4 and the pseudo-acoustic echo that is the output of the variable coefficient filter 3 are input to the subtraction circuit 5, the acoustic echo 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. The reception signal is input to the reception signal input register 8 in the variable coefficient filter 3, and the sum of products 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 sum of products circuit 10. Sum of products 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 to four-wire converter in the case of a long-distance telephone line, and to a speaker and a microphone in the case of a public telephone 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 = hT k (1) で表される。但し、 h=[h1 ,h2 ,・・・,hn T (2) x=[xk-1 ,・・・,xk-n T T :べクトルの転置 である。
Assuming that the signal propagation characteristic of the acoustic echo path is linear and represented by an FIR type digital filter, if the impulse response h (t) and the input received signal x (t) are used, the sampling time interval is Is T and the acoustic echo y k at time kT is represented by y k = h T x k (1). However, h = [h 1 , h 2 , ..., H n ] T (2) x = [x k−1 , ..., x kn ] T T : transposition of the vector.

【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 T k (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 T k (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
k 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 h
If s k , the estimated value ys k of y k is given by ys k = hs k T x k (3). In the acoustic echo canceller, when there is a voice signal in the reception signal input terminal 1 and there is no voice signal in the transmission signal input terminal 4 and only acoustic echo exists, the echo elimination operation is performed as an adaptive operation state. This adaptive motion algorithm is generally a learning identification method (Atsuhiko Noda, Jinichi Nagumo: “System Learning Identification Method” Measurement and Control, 7, 9, pp.597-605 (1
968)) is adopted. The sequential correction of hs k by the learning identification method is performed by hs k + 1 = hs k + α (x k e k ) / x k T x k (4). However, e k = y k −y s k , 0 <α ≦ 1 (5), and e k is called residual acoustic echo. Such a calculation operation is processed in the coefficient correction amount calculation circuit 7. The contents of the pseudo impulse response register 9 include the variable coefficient series h.
s k is stored. α is a coefficient update gain for determining the sensitivity of estimation, and the closer it is to 1.0, the larger the amount of correction can be given, and high-speed acoustic echo removal can be performed. However, in actual use, near-end noise can be reduced. It is necessary to change the setting depending on the line status. The fact that the coefficient update gain α is currently determined depends on an empirical rule. There are also ones that variably control the coefficient updating gain α depending on the magnitude of the residual acoustic echo and ones that are set according to the indoor characteristics (for example, Shoji Makino and Nobuo Koizumi: “The adaptive characteristics of echo cancellers in the room sound field. Regarding improvement ”, J. A., J71-A, 12, pp.2212-2214 (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。一例を図5に
示す。図中aは係数更新利得αを0.5に固定した場
合、同図bは室内特性に合わせた係数更新利得αの設定
を行った場合の音響反響除去特性である。係数更新利得
αの設定の仕方によって消去特性に大きな差が現れる事
が判る。固定利得とした場合には高速性は維持できる
が、動作安定性が失われる危険性が増大する。又、室内
特性に合わせて設定した場合には動作安定性は向上する
ものの適応速度が著しく劣化してしまう。この様に係数
更新利得αの設定には、相反する条件を満たさなければ
ならないという問題点があった。
The method of least squares (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 the equation (5), if the range of α is between 0 and 1, some performance can be obtained, but the convergence speed and the saturation echo removal amount differ due to the difference in the value. An example is shown in FIG. In the figure, a is the acoustic echo removal characteristic when the coefficient update gain α is fixed at 0.5, and b in the figure is the acoustic echo removal characteristic when the coefficient update gain α is set according to the indoor characteristic. It can be seen that a large difference appears in the erasing characteristic depending on the setting method of the coefficient update gain α. When a fixed gain is used, high speed performance can be maintained, but the risk of loss of operational stability increases. Further, if the setting is made according to the indoor characteristics, the operation stability is improved, but the adaptation speed is significantly deteriorated. As described above, the setting of the coefficient update gain α has a problem that conflicting conditions must be satisfied.

【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こ
とを目的とする。
The present invention has been made in view of the above points, eliminates the above problems, and is excellent in high speed and operational stability.
An object of the present invention is to provide an acoustic echo canceller having high adaptive performance and performing acoustic control while always maintaining a large amount of acoustic echo cancellation.

【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、該受話信号入力端子の受話信号を入力とする擬似
インパルス応答レジスタを持つ可変係数ディジタルフィ
ルタと、該受話信号出力端子から音響反響経路を介して
該送話信号入力端子に入力される受話信号の音響反響成
分から該可変係数ディ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様
な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、該擬似インパルス応答レ
ジスタをN個のブロックに分割し、各ブロック毎の係数
累積加算平均電力の代表変動率を算出し、その算出され
た代表変動率により各ブロック毎の係数更新利得が変更
される音響反響除去装置を提供する。
SUMMARY OF THE INVENTION The present invention is to solve these problems and comprises a reception signal input terminal, a reception signal output terminal, a transmission signal input terminal, and a transmission signal output terminal. A variable coefficient digital filter having a pseudo impulse response register that receives the reception signal of the reception signal input terminal, and the sound of the reception signal input from the reception signal output terminal to the transmission signal input terminal through an acoustic echo path. In an acoustic echo canceling apparatus in which a coefficient sequence is sequentially updated by a coefficient correction amount arithmetic circuit that minimizes a residual signal obtained by subtracting a pseudo acoustic echo generated by the variable coefficient digital filter from an echo component, The pseudo impulse response register is divided into N blocks, the representative variation rate of the coefficient cumulative addition average power for each block is calculated, and the representative variation rate is calculated. To provide an acoustic echo removal device coefficient update gain for each block is changed.

【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 estimation operation are ensured by the above means, the reverberation component is hardly mixed into the communication line, the deterioration of the communication voice sound quality is prevented, and the call itself. It is possible to reduce the risk of howling that would otherwise be impossible, and high-quality acoustic control becomes possible.

【0010】[0010]

【実施例】本発明の実施例について、図面を用いて説明
を行う。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で示されるように、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11、
ブロック平均累積電力演算回路12、ブロック電力変動
率算出回路13、そして、ブロック係数更新利得選択回
路14を追加した構成である。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、各ブロック
独立の平均累積加算平均電力を算出する該ブロック平均
累積電力演算回路12と、各ブロック独立の平均累積加
算平均電力の変動率を算出する該ブロック電力変動率算
出回路13と、各ブロックの平均累積加算平均電力の変
動率に対応した該ブロック係数更新利得選択回路14に
記憶された0.0から1.0の範囲の各ブロックに対応
した係数更新利得αを毎回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、この値を用いて式(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。該ブロック平均累積電力演算回路12において、該
擬似インパルス応答レジスタに格納された可変係数の累
積加算平均電力の各ブロック毎の平均電力PBNを式
(6)により算出する。各ブロックのサイズは等間隔で
も不当間隔であっても良い。
Embodiments of the present invention will be described with reference to the drawings. FIG. 1 is a block diagram showing the configuration of the acoustic echo canceller of the present invention. As shown in FIG. 1, according to the present invention, the conventional reception signal input terminals 1, 3 reception signal output terminals 2, variable coefficient digital filter 3, transmission signal input terminal 4, subtraction circuit 5, transmission signal output terminal 6, Coefficient correction amount calculation circuit 7,
A cumulative addition / average power calculation circuit 11 is provided in a device having the same configuration as the acoustic echo canceling device that employs a learning identification method as an adaptive algorithm, which is composed of a reception signal input register 8, a pseudo impulse response register 9, and a product-sum calculation circuit 10. ,
This is a configuration in which a block average cumulative power calculation circuit 12, a block power fluctuation rate calculation circuit 13, and a block coefficient update gain selection circuit 14 are added. The reception signal input terminal 1
, The reception signal output terminal 2 and the transmission signal input terminal 4
And the transmission signal output terminal 6 and the pseudo impulse response register 9 that receives the reception signal of the reception signal input terminal 1
And the variable coefficient digital filter 3 generated from the acoustic echo component of the received signal input to the transmitted signal input terminal 4 from the received signal output terminal 2 via the acoustic echo path. In the acoustic echo canceller in which the coefficient sequence is sequentially updated by the coefficient correction amount calculation circuit 7 so as to minimize the residual signal obtained by subtracting the pseudo acoustic echo, the pseudo impulse response register 9
The cumulative average power calculation circuit 11 for calculating the cumulative average power of each variable coefficient alone stored in, and the block average cumulative power calculation circuit 12 for calculating the average cumulative average power of each block, It is stored in the block power fluctuation rate calculation circuit 13 that calculates the fluctuation rate of the block-independent average cumulative addition average power, and in the block coefficient update gain selection circuit 14 that corresponds to the fluctuation rate of the average cumulative addition average power of each block. The coefficient update gain α corresponding to each block in the range of 0.0 to 1.0 is selected each time, the selected value is sent to the coefficient correction amount calculation circuit 7, and this value is used to correct by the equation (4). An acoustic echo canceller characterized by calculating a quantity. In the block average cumulative power calculation circuit 12, the average power P BN of each block of the cumulative addition average power of the variable coefficient stored in the pseudo impulse response register is calculated by the equation (6). The size of each block may be equally spaced or may be unequally spaced.

【0011】 PB1=(hs0 2+・・・ +hsm1-1 2 )/m1 PB2=(hsm1 2 +・・・ +hsm2-1 2 )/(m2−m1) (6) | PBN=(hsmN 2 +・・・ +hsL-1 2)/(L−mN) Lは可変係数の総数、m1、m2、…、mNは分割する
係数ナンバー、そして、Nは分割ブロックの総数であ
る。ここでは評価値としてブロック毎平均を用いている
が、そのブロックを代表できる値ならば、この値ではな
くとも例えば係数累積加算平均電力の最大値、最小値で
あってもよい。該ブロック平均累積電力演算回路12の
構成を図2に示す。各平均電力PBNを用いて該ブロック
電力変動率算出回路13では、式(7)を用いて算出を
行う。
P B1 = (hs 0 2 + ... + hs m1-1 2 ) / m1 P B2 = (hs m1 2 + ... + hs m2-1 2 ) / (m2-m1) (6) | P BN = total (hs mN 2 + ··· + hs L-1 2) / (L-mN) L is variable coefficient, m1, m @ 2, ..., mN coefficient number divides Then, the total number of N is divided block Is. Here, the average for each block is used as the evaluation value, but if it is a value that can represent that block, it may be the maximum value or the minimum value of the coefficient cumulative addition average power instead of this value. The configuration of the block average cumulative power calculation circuit 12 is shown in FIG. The block power fluctuation rate calculation circuit 13 uses each average power P BN to perform the calculation using equation (7).

【0012】 dPBN(k+1 )=|PBN(k+1 )−PBN(K )|/PBN(K ) (7) 式(7)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、dPBN(k+1 )とdPBN(K )との間での差分
値を用いても本発明は有効に機能する。該ブロック係数
更新利得選択回路14では、各ブロックの代表変動率d
BN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3にこの概念図を示す。この時、
ブロック代表評価値の変動率が一定以下になったと検出
した場合には、そのブロック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、演算誤差による誤動作発生の危険性を
低く抑える事ができる。図4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と室内特性を考慮にいれたモデルの結果c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[dB]
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。
DP BN (k + 1) = | P BN (k + 1) −P BN (K) | / P BN (K) (7) The fluctuation rate shown in the equation (7) is an example, and is determined. It is not a calculation method. For example, the numerator may be the present value instead of the past value. The present invention also works effectively by using the difference value between dP BN (k + 1) and dP BN (K). In the block coefficient update gain selection circuit 14, the representative variation rate d of each block
Compare the value of P BN with several thresholds interpolated in the circuit,
An operation of selecting a coefficient update gain corresponding to a suitable threshold range is performed. FIG. 3 shows this conceptual diagram. This time,
When it is detected that the variation rate of the block representative evaluation value becomes equal to or less than a certain value, the coefficient correction amount calculation circuit 7 of the block is detected.
By setting the correction amount calculation and the update calculation not to be performed, the calculation amount can be reduced, the load on the hardware can be reduced, and the risk of malfunction due to a calculation error can be reduced. FIG. 4 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 and the result c of the model in consideration of the indoor characteristics are shown in the same figure. The vertical axis represents the amount of acoustic echo cancellation, and the horizontal axis represents time. The maximum setting value of the coefficient update gain of the model according to the present invention is 1.0 and the minimum setting value is 0.05. The response speed in the initial cancellation transient region is 30 [dB] for the acoustic echo cancellation.
It can be seen that the model according to the present invention is superior to the model according to the present invention by about 2 times. Since the coefficient update gain is gradually set to a small value, it is not affected by disturbance.

【0013】[0013]

【発明の効果】以上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。 (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。
As described above, according to the present invention, the following excellent effects are expected. (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 risk of howling can be suppressed to a low level. (2) In the adaptive operation process, a variable coefficient that does not need to be updated is generated. This reduces the amount of calculation,
The burden on the hardware can be reduced.

【0014】(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 coefficient is not updated more than necessary, an arithmetic error or malfunction is less likely to occur when the digital signal processor or the like is used.

【図面の簡単な説明】[Brief description of 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 echo canceller according to the present invention.

【図2】 本発明の説明に用いたブロック平均累積電力
演算回路の一構成を示すブロック図である。
FIG. 2 is a block diagram showing a configuration of a block average cumulative power arithmetic circuit used for explaining the present invention.

【図3】 状態判定制御部の概念を示すブロック図であ
る。
FIG. 3 is a block diagram showing a concept of a state determination control unit.

【図4】 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. 4 is a diagram showing an example of acoustic echo cancellation characteristics when white noise according to the present invention is used as a reference input.

【図5】 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. 5 is a diagram showing an example of acoustic echo canceling characteristics by a conventional method when the white noise according to the present invention used in the present description is used as a reference input.

【図6】 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. 6 is a block diagram showing an example of a basic configuration of an acoustic echo canceller using a conventional 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 ブロック平均累積電力演算回路 13 ブロック電力変動率算出回路 14 ブロック係数更新利得選択回路 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 calculation circuit 8 reception signal input register 9 pseudo impulse response register 10 sum of products calculation circuit 11 Cumulative average power calculation circuit 12 block average cumulative power calculation circuit 13 block power fluctuation rate calculation circuit 14 block coefficient update gain selection circuit

Claims (1)

【特許請求の範囲】[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、該受話
信号入力端子の受話信号を入力とする擬似インパルス応
答レジスタを持つ可変係数ディジタルフィルタと、該受
話信号出力端子から音響反響経路を介して該送話信号入
力端子に入力される受話信号の音響反響成分から該可変
係数ディ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様な係数修正量
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、該擬似インパルス応答レジスタをN個
のブロックに分割し、各ブロック毎の係数累積加算平均
電力の代表変動率を算出し、その算出された代表変動率
により各ブロック毎の係数更新利得が変更されることを
特徴とする音響反響除去装置。
1. A variable 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 to which the reception signal of the reception signal input terminal is input. A coefficient digital filter and a pseudo acoustic echo generated by the variable coefficient digital filter are subtracted from the acoustic echo component of the reception signal input to the transmission signal input terminal from the reception signal output terminal through the acoustic echo path. In an acoustic echo canceller in which a coefficient series is sequentially updated by a coefficient correction amount calculation circuit that minimizes a residual signal to be obtained, the pseudo impulse response register is divided into N blocks, and coefficient accumulation for each block is performed. An acoustic echo characterized by calculating the representative fluctuation rate of the average power and changing the coefficient update gain for each block by the calculated representative fluctuation rate. Removal device.
JP26363894A 1994-10-27 1994-10-27 Acoustic echo canceller Expired - Fee Related JP3217618B2 (en)

Priority Applications (1)

Application Number Priority Date Filing Date Title
JP26363894A JP3217618B2 (en) 1994-10-27 1994-10-27 Acoustic echo canceller

Applications Claiming Priority (1)

Application Number Priority Date Filing Date Title
JP26363894A JP3217618B2 (en) 1994-10-27 1994-10-27 Acoustic echo canceller

Publications (2)

Publication Number Publication Date
JPH08125589A true JPH08125589A (en) 1996-05-17
JP3217618B2 JP3217618B2 (en) 2001-10-09

Family

ID=17392279

Family Applications (1)

Application Number Title Priority Date Filing Date
JP26363894A Expired - Fee Related JP3217618B2 (en) 1994-10-27 1994-10-27 Acoustic echo canceller

Country Status (1)

Country Link
JP (1) JP3217618B2 (en)

Cited By (1)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
KR100337535B1 (en) * 1997-05-12 2002-07-18 사와무라 시코 Echo canceler having adaptive filter and method of reinitiating coefficients of adaptive filter

Cited By (1)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
KR100337535B1 (en) * 1997-05-12 2002-07-18 사와무라 시코 Echo canceler having adaptive filter and method of reinitiating coefficients of adaptive filter

Also Published As

Publication number Publication date
JP3217618B2 (en) 2001-10-09

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